| 00:03.45 | *** join/#asterisk Hmmhesays (n=Neg@24-117-131-41.cpe.cableone.net) |
| 00:07.07 | Qwell | garreel: Convert it to 8k mono |
| 00:07.45 | crudi | i tried deleting the voice menu, but now it just rings and i dont get anything |
| 00:15.28 | crudi | i dont see anything in extensions that would do that |
| 00:17.30 | *** join/#asterisk bobesponja (n=pat@bas75-1-82-235-231-205.fbx.proxad.net) |
| 00:19.13 | bobesponja | <PROTECTED> |
| 00:19.55 | JT | yep it works fine |
| 00:25.02 | crudi | my verbose output says "s" is an invalid extension |
| 00:25.07 | crudi | here is my trunk config |
| 00:25.08 | crudi | http://www.pastebin.ca/419040 |
| 00:25.24 | crudi | thats what the gui put |
| 00:26.22 | Hmmhesays | working here |
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| 00:38.42 | crudi | also, when i call VM it doesn't respond when i push any options |
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| 00:42.06 | garreel | any resource on fax detection on outbound calls? |
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| 00:54.39 | ber_ | question about AGI, anyone here have experience with it? |
| 00:55.35 | gambolputty | some |
| 00:56.31 | ber_ | I am doing a get_data request |
| 00:56.36 | ber_ | inside of a loop |
| 00:56.42 | ber_ | the first request works fine |
| 00:56.50 | ber_ | and returns the right digits |
| 00:57.05 | ber_ | but on the second iteration of the loop, get_data is bypassed and the rest of the loop is executed |
| 00:57.12 | gambolputty | get data to an external program? |
| 00:57.30 | ber_ | yes |
| 00:57.35 | ber_ | i am using the Perl AGI module |
| 00:57.41 | gambolputty | maybe the loop is programmed wrong |
| 00:57.49 | ber_ | possible, the things right after it execute |
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| 00:57.58 | ber_ | its just a while(1) |
| 00:58.13 | ber_ | like right after it i have a playback('somewavfile') |
| 00:58.15 | gambolputty | look at that then |
| 00:58.27 | gambolputty | or use a for/next style iteration loop |
| 01:00.53 | mmartinn | show us with a pastebin |
| 01:03.05 | garreel | do I need spansp to make fax detection (not receiving, only detection) working? |
| 01:04.29 | ber_ | yeah its something wrogn wtih my code |
| 01:05.07 | mmartinn | ber_: You're checking that get data is not returning -1? |
| 01:05.21 | mmartinn | ber_: that may break your loop condition |
| 01:07.23 | crudi | any idea why my phone wouldnt be accepting digits in my VM? |
| 01:09.16 | ber_ | i am trying to isolate it further |
| 01:09.28 | ber_ | when i just have a while loop with only 2 things a get_data and a wavefile play |
| 01:09.30 | ber_ | it works |
| 01:09.35 | ber_ | asks for get_data each time |
| 01:09.42 | ber_ | but when i put it in the big program |
| 01:09.49 | ber_ | asterisk CLI says it is executing get_data |
| 01:09.56 | ber_ | but i never get a prompt to enter a value |
| 01:10.01 | mmartinn | ber_: show us |
| 01:10.02 | mmartinn | ~pb |
| 01:10.18 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 01:10.18 | ber_ | ok here is the loop that works |
| 01:12.08 | riddlebox | where are the voicemails located on the system? |
| 01:13.27 | ber_ | http://www.pastebin.ca/419082 |
| 01:14.45 | ber_ | i just added what the CLI output is |
| 01:14.48 | ber_ | for the 2nd loop |
| 01:16.09 | mmartinn | ber_: I don't see anything off the top of my head, but i'm not a perl person |
| 01:16.39 | ber_ | yeah the thing i dont get is the loop is still going |
| 01:16.45 | ber_ | and the asterisk CLI says it is executing the get_data |
| 01:16.58 | ber_ | but no prompt and no wavefile play |
| 01:17.14 | ber_ | anyways thanks for looking i will figure it out |
| 01:17.44 | mmartinn | ber_: It sounds a like perl thing; like execution is interrupted |
| 01:18.15 | ber_ | i guess could be a bug in the perl module |
| 01:18.42 | ber_ | i can probably simplify the section related to IVR |
| 01:18.52 | ber_ | and move all the other code out to subroutines |
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| 01:43.29 | ber_ | it was a perl thing |
| 01:43.33 | ber_ | brute forced it out |
| 01:43.40 | ber_ | does perl have a debugger |
| 01:45.50 | Tene | perl -d |
| 01:49.55 | ber_ | can you attach to a runnig process with that |
| 01:50.24 | Tene | Unsure. |
| 01:50.35 | Tene | perldoc perldebug |
| 01:51.14 | ber_ | ok ill google 2 |
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| 04:13.30 | bsd_tech | ok found a issue with the configure in the addons in asterisk-oh3232 |
| 04:13.44 | bsd_tech | its not including /usr/local/include |
| 04:13.55 | bsd_tech | so it wont build on bsd |
| 04:14.11 | bsd_tech | we have everythign else working and yes this is 1.4 |
| 04:15.40 | bsd_tech | how can I fix the configure to include the /usr/local/include |
| 04:16.20 | russellb | well, if it was in the default search path for your compiler, this would not be a problem. |
| 04:18.16 | russellb | what is oh323 dependent on? |
| 04:18.46 | coppice | this is a common source of annoyance. most systems that use /usr and /usr/local are not set up by default to include /usr/local |
| 04:18.47 | bsd_tech | its in the main make file but the configure in asterisk-oh323 seems not to read it |
| 04:19.21 | bsd_tech | checking for ANSI C header files... (cached) yes |
| 04:19.21 | bsd_tech | checking for /usr/include/asterisk/channel.h... no |
| 04:19.21 | bsd_tech | configure: error: No asterisk installation found |
| 04:19.37 | russellb | yeah, we just add /usr/local if it is freebsd by default |
| 04:19.42 | bsd_tech | checking for /usr/include/asterisk/channel.h... no shold also ask in /usr/local/include |
| 04:20.09 | bsd_tech | checking for /usr/include/asterisk/channel.h... it shold also checkin /usr/local/include |
| 04:20.19 | russellb | right. |
| 04:20.24 | bsd_tech | sorry tired did alot of coding and patching today |
| 04:21.20 | bsd_tech | this is the last thing I have to fix os 1.4 on bsd |
| 04:21.30 | bsd_tech | os /on |
| 04:23.04 | bsd_tech | is there a way to fix this ? |
| 04:23.17 | russellb | of course |
| 04:23.31 | bsd_tech | how and where |
| 04:23.52 | bsd_tech | like I said been along day making patches |
| 04:24.37 | russellb | well, to hack it for yourself, just search for that path in the configure script and replace it. |
| 04:24.59 | russellb | to do a "real" fix, a new test to look for the asteirsk headers needs to be written |
| 04:25.11 | russellb | and it shouldn't be in the oh323 configure script, it should be in the top level one |
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| 04:26.40 | bsd_tech | includedir='${prefix}/include' |
| 04:26.40 | bsd_tech | oldincludedir='/usr/include' |
| 04:26.44 | bsd_tech | should find it |
| 04:26.53 | bsd_tech | but it does not |
| 04:27.15 | russellb | what? |
| 04:27.37 | bsd_tech | prefix=${prefix} |
| 04:27.52 | bsd_tech | hmmm |
| 04:28.47 | l0rdr0ck | hey bsd_tech. |
| 04:28.56 | bsd_tech | hey |
| 04:29.00 | l0rdr0ck | with your help i won that bounty |
| 04:29.02 | bsd_tech | you fell off the other day |
| 04:29.06 | bsd_tech | cool |
| 04:29.09 | l0rdr0ck | yup. |
| 04:29.21 | l0rdr0ck | man i was working on other stuff with the box. sorry |
| 04:29.37 | bsd_tech | its ok |
| 04:29.44 | bsd_tech | I just pm you back |
| 04:30.19 | bsd_tech | I am working on the last bit of asterisk 1.4 on bsd now |
| 04:30.25 | bsd_tech | the asterisk-addons |
| 04:36.09 | bsd_tech | work on this tomarrow night guys |
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| 05:19.23 | haiger | hi everyone |
| 05:20.25 | haiger | i am very n00be to astreisk, just wanted to have a general idea, what hardware do i have to have to run asterisks, i am very intrested |
| 05:21.59 | florz | haiger: A Computer would be a good start. |
| 05:22.02 | Tene | haiger: you don't need much at all. |
| 05:22.43 | haiger | then i guess i am ready :) |
| 05:22.50 | florz | *g* |
| 05:23.17 | haiger | i am running debian etch, i will install astreisk zaptel |
| 05:23.38 | florz | haiger: Erm, and a network adapter probably wouldn't be bad, if you don't wanna to talk to yourself only =:-) |
| 05:24.07 | florz | -to |
| 05:24.37 | haiger | Question:for the phone part, i don't have ATA, is there is a way arround this, like use my headset for example |
| 05:25.09 | Tene | haiger: Sure! |
| 05:25.12 | l0rdr0ck | question: anyone working with Astribank on OS X? |
| 05:26.20 | haiger | i have a linksys ATA from vonage when i use to have vonage , but it's locked to vonage :( |
| 05:26.46 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 05:26.46 | *** mode/#asterisk [+o mog] by ChanServ |
| 05:28.35 | Tene | haiger: you can use your headset just fine! |
| 05:31.17 | haiger | darn i can't find asterisk-zaptel for debian |
| 05:32.03 | carrar | compile it? |
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| 05:38.57 | haiger | i just did apt-get install asterisk and it installing right now |
| 05:45.55 | haiger | Tene can i get help...Asterisk not yet configured. Edit /etc/default/asterisk first |
| 05:46.08 | Tene | haiger: look at files in /etc/asterisk/ |
| 05:47.21 | haiger | Tene: what am i looking for please |
| 05:47.37 | Tene | haiger: that's where asterisk stores its configuration files |
| 05:48.08 | haiger | can you help me please , i am at that folder and i am lost |
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| 05:49.36 | Tene | haiger: I'm sorry, I can't right now. I'm too busy right now to teach you how to use asterisk. |
| 05:49.47 | Tene | There's some good documentation and tutorials out on the internet, I think. |
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| 05:51.12 | haiger | thanks, anyway ;) |
| 05:51.16 | MetztliXictli | hi |
| 05:51.34 | MetztliXictli | speak spanish= |
| 05:52.07 | haiger | no apla englisa sir |
| 05:55.16 | [TK]D-Fender | <PROTECTED> |
| 05:55.32 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:57.44 | MetztliXictli | i want test asteris to cabina telephony |
| 05:57.51 | MetztliXictli | *asterisk |
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| 06:39.06 | andresmujica | hi i'm connecting an asterisk box to a panasonic one. i' m having trouble with hangup detection. anyone can help me? |
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| 06:51.29 | Qwell | http://www.google.com/tisp/ |
| 06:51.39 | Qwell | That certainly explains why they've been buying all that dark fiber |
| 06:52.38 | Tene | Qwell: look at the date |
| 06:52.56 | BBHoss | haha |
| 06:53.06 | Qwell | oh, really, you mean it isn't possible? |
| 06:53.10 | BBHoss | thats even better than google paper |
| 06:53.18 | BBHoss | it has been pulled before |
| 06:53.28 | BBHoss | you just have to put ends on the fiber |
| 06:53.33 | Tene | Qwell: no, I'm just asking if archive.org saw that page before today. |
| 06:54.57 | BBHoss | new to me |
| 06:55.26 | Tene | google has a history of "releasing" gags on this date |
| 06:56.25 | Tene | Qwell: also, look at the 'get started now' instructions. |
| 06:58.15 | mog | lol sarcasm is not working for qwell.... |
| 06:58.31 | Qwell | I mean, come on, how much more blatent do I need to be? :P |
| 06:58.58 | droops | Qwell, i think the toliet fiber is a joke |
| 06:59.14 | Qwell | repeatedly |
| 06:59.16 | mog | lol |
| 06:59.22 | mog | poor qwell |
| 06:59.55 | Qwell | No new RFC yet :( |
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| 07:02.41 | Tene | Qwell: ahh. Sorry. The social parts of my brain turned off a few hours ago. |
| 07:02.50 | Qwell | Tene: :p |
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| 07:08.00 | andresmujica | would it be a hoax?? send the fiber trhough the toilet??? |
| 07:08.24 | Qwell | yeah, I'm just gonna go ahead and go to bed |
| 07:08.29 | andresmujica | ohh that's right |
| 07:08.33 | andresmujica | april fool' s day |
| 07:08.50 | andresmujica | last year was asterisk for win2k |
| 07:09.31 | fbffff | and how is that funny? |
| 07:10.17 | andresmujica | the press release said that the linux version would be droped... and it has some fun remarks about win2k stability etc... |
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| 07:10.29 | andresmujica | had.. |
| 07:12.04 | Qwell | I'm actually kinda surprised the google folks didn't think of the biggest problem with using fiber for that |
| 07:12.23 | Qwell | the bottoms of toilets have a very sharp curve, which would break fiber instantly |
| 07:12.54 | Qwell | it isn't a technically sound idea, heh |
| 07:14.15 | droops | but the whole reference to fiber and toilets sounds fun |
| 07:14.31 | andresmujica | yeapp... |
| 07:14.41 | fbffff | What are the system requirements? |
| 07:14.45 | fbffff | Windows XP/Vista (Mac and Linux support coming soon) |
| 07:14.45 | fbffff | Internet Explorer 6.0+ or Firefox 1.5+ with the Google Toolbar |
| 07:14.45 | fbffff | Round-front or elongated toilet providing at least 1.0 gallons per flush |
| 07:14.45 | fbffff | Use of automatic toilet bowl cleaners is not recommended |
| 07:14.46 | fbffff | hah |
| 07:14.51 | fbffff | only windows is supported |
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| 07:40.15 | andresmujica | ok |
| 07:40.32 | andresmujica | i' ve managed to get my panasonic and asterisk box working nice together!!!!! |
| 07:40.41 | Tene | yay! |
| 07:40.42 | Tene | :D |
| 07:40.51 | andresmujica | with hangup detection from fxo to fxo to fxs and whatever!!! |
| 07:40.55 | andresmujica | yupa!!!! |
| 07:41.15 | andresmujica | well it was about time ... 2:41 am :( |
| 07:41.20 | l0rdr0ck | congrats |
| 07:41.41 | andresmujica | thks!! |
| 07:41.58 | andresmujica | no where i can find one ogf those toilet kits!??? |
| 07:42.01 | andresmujica | now |
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| 08:30.18 | stony | hi |
| 08:36.53 | stony | every time i load the hfcmulti module, i get 4 ports that are unusable |
| 08:37.08 | stony | if i unload the module and reload it, then the init of the card is fine |
| 08:37.39 | stony | but the first 4 ports that don't work are still listed and the 8 ports of the card are begining from 5th position |
| 08:37.58 | stony | so i have 1-4 old, not working, ports and 5-12 "new", working ports |
| 08:38.03 | stony | anyone ever had this problem ? |
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| 10:34.55 | *** part/#asterisk andresmujica (n=andresmu@200.75.38.138) |
| 10:54.18 | *** join/#asterisk Ast001 (n=uros@212.200.223.176) |
| 10:58.22 | *** join/#asterisk stony (n=steinche@p5B150D5B.dip0.t-ipconnect.de) |
| 11:03.04 | *** join/#asterisk ligoban (n=pseudor@193.37.156.146) |
| 11:10.55 | *** join/#asterisk bartpbx (n=bartpbx@dslb-084-059-130-117.pools.arcor-ip.net) |
| 11:11.00 | bartpbx | hello |
| 11:11.13 | bartpbx | anyone out there using zap and misdn for data calls? |
| 11:15.56 | *** join/#asterisk TEOX (n=TEX@203-184-35-128.jetbuster.co.nz) |
| 11:16.42 | TEOX | using a grandstream 100, on asterisknow and getting mic feedback when connecting to pstn |
| 11:16.51 | TEOX | like i can hear myself very loud |
| 11:17.05 | TEOX | and can even pickup noise in the background |
| 11:17.12 | TEOX | hardware or a software issue? |
| 11:18.21 | TEOX | its like over sensitive mic |
| 11:18.22 | tzafrir_laptop | bartpbx, "data calls"? calls are generally audio. Do you mean something like "data links"? BRI or PRI? |
| 11:18.36 | *** join/#asterisk Dovid (n=Dovid@85.159.160.207) |
| 11:18.43 | Dovid | anyone see these links ? |
| 11:19.18 | robin_sz | tzafrir, misdn presumably means chan_mISDN ... |
| 11:19.36 | Dovid | http://www.google.com/tisp/press.html |
| 11:19.37 | Dovid | http://www.google.com/tisp/install.html |
| 11:20.13 | robin_sz | Dovid, I was a beta tester for that, works OK, assuming hte initial connection goes well |
| 11:20.20 | Dovid | haha |
| 11:20.29 | Dovid | robin_sz: It's an april fools joke |
| 11:20.58 | Dovid | didnt know that google did those ;) |
| 11:21.38 | robin_sz | Dovid, as a hint ... if trying to do an april fools joke on a channel, it helps to have a few people join in and sucker the rest |
| 11:21.52 | Dovid | tooo tired. |
| 11:21.53 | Dovid | lol |
| 11:22.12 | Dovid | robin_sz: will try it later. |
| 11:22.13 | Dovid | lol |
| 11:22.17 | robin_sz | its the only joke ive seen today |
| 11:22.30 | Dovid | and i fell for it for abbout 2 minutes. |
| 11:22.30 | robin_sz | traditionally, you are supposed to stop at midday anyway |
| 11:22.35 | Dovid | ww.woot.com |
| 11:22.38 | Dovid | www.woot.com |
| 11:23.00 | Dovid | and every year some one on the biz list says that * 2.0 was released. every year. so this year i sent it out to get it over with |
| 11:23.20 | robin_sz | heh |
| 11:23.42 | JT | tzafrir_laptop: you can make data calls on isdn bri |
| 11:24.00 | JT | setup flags it as DATA not 3K1AUDIO or SPEECH |
| 11:24.04 | tzafrir | never tried, actually |
| 11:24.07 | JT | telcos usually charge more |
| 11:24.30 | JT | old isdn internet connections.... 64 or 128kbit/s :P |
| 11:25.20 | tzafrir | well, we probably have a better adsl connection |
| 11:25.26 | *** join/#asterisk voltagex (n=voltagex@124-254-124-5-dsl.ispone.net.au) |
| 11:25.34 | JT | yes adsl is generally better |
| 11:25.34 | tzafrir | higher latency, but still |
| 11:25.38 | JT | except you can't call people |
| 11:26.50 | voltagex | hi, anyway to set a phone as unavailable/dnd if the phone doesn't have a button for that? |
| 11:27.13 | Dovid | voltagex: u can set it in the asterisk db |
| 11:27.44 | voltagex | Dovid: well I want to be able to turn it on and off, like by pressing # |
| 11:27.53 | voltagex | so that while I'm sleeping, calls go to voicemail |
| 11:28.09 | Dovid | yes. |
| 11:28.15 | JT | setup an extension to set it |
| 11:28.20 | Dovid | u can create an extension that will set the phone to DND |
| 11:28.24 | voltagex | oh |
| 11:28.37 | voltagex | and how do I tell it which voicemailbox to use? |
| 11:29.49 | robin_sz | in sip.conf |
| 11:30.22 | voltagex | got it |
| 11:30.22 | voltagex | http://etel.wiki.oreilly.com/wiki/index.php/Do_Not_Disturb |
| 11:32.00 | robin_sz | are you using the 'standard extension' macro? |
| 11:32.19 | robin_sz | set the vm extension in there |
| 11:32.39 | robin_sz | and set vmexten=<number> in sip.conf. job done |
| 11:33.06 | *** part/#asterisk TEOX (n=TEX@203-184-35-128.jetbuster.co.nz) |
| 11:33.07 | voltagex | robin_sz: what's the standard extension macro? |
| 11:33.23 | robin_sz | you hvae the sample configs? |
| 11:33.31 | voltagex | yes |
| 11:34.00 | robin_sz | look in extensions.conf |
| 11:34.09 | voltagex | ok |
| 11:36.30 | robin_sz | macro-stdexten // hmm i think i might have tweaked mine ... |
| 11:36.56 | voltagex | in that oreilly example, how can I change that from-internal context to work for all of my calls, internal and externa,? |
| 11:37.06 | voltagex | external* |
| 11:38.35 | Dovid | robin_sz: Here is another |
| 11:38.36 | Dovid | http://blog.roam4free.ie/jajah-buys-vonage/ |
| 11:39.40 | *** join/#asterisk shay|work (i=shay@unaffiliated/shay) |
| 11:39.43 | shay|work | hello folks |
| 11:39.55 | Dovid | helo shay |
| 11:40.25 | shay|work | I just got a Digium TDM400 card, with four RJ-11C connectors |
| 11:40.33 | shay|work | s/connectors/ports |
| 11:40.43 | shay|work | how can I know which port is FXO and which one is FXS? |
| 11:40.44 | Dovid | ok |
| 11:41.12 | Dovid | the red ones are FXO green ones are FXS |
| 11:41.22 | Dovid | also once installed if you do ztcfg -vvvv |
| 11:41.24 | shay|work | the leds you mean? |
| 11:41.24 | Dovid | it will tell u |
| 11:41.33 | Dovid | yes |
| 11:41.43 | shay|work | then I guess that I got four FXS ports.. |
| 11:41.59 | shay|work | does this means that I cannot connect the phone line to that card? |
| 11:42.01 | Dovid | nno - sorry the LED's tell u that it's working |
| 11:42.14 | Dovid | the daugher boards on the card its self. |
| 11:42.18 | Dovid | are they red ? green ? |
| 11:42.20 | *** join/#asterisk Ebola (n=Ebola@host86-136-190-11.range86-136.btcentralplus.com) |
| 11:42.46 | shay|work | you mean the small cards connected to the big card in one side? |
| 11:42.54 | Dovid | yes |
| 11:42.57 | Dovid | what color are they ? |
| 11:43.03 | shay|work | they are four gren small cards |
| 11:43.14 | Dovid | yes. then u have all fxs |
| 11:43.35 | shay|work | so, does this means that I cannot connect the phone line to that card? |
| 11:43.46 | shay|work | only analog phones? |
| 11:43.47 | Dovid | brb phone |
| 11:43.52 | Dovid | correct |
| 11:43.52 | shay|work | sure, tyt |
| 11:43.57 | shay|work | Dovid: thanks a lot |
| 11:45.00 | ligoban | Dovid: Excuse me, I have a question about installation * addons-1.4.0 |
| 11:45.05 | tzafrir | and you need power for them... |
| 11:46.19 | Dovid | libogan: never used 1.4.x but what is ur wuestion ? |
| 11:46.43 | Dovid | shady: where did u order the card from and what did u ask for when u orderd it ? |
| 11:48.45 | shay|work | tzafrir: hey :-) |
| 11:49.23 | shay|work | Dovid: was that question derected to me? |
| 11:49.47 | Dovid | yes |
| 11:49.58 | Dovid | hence the shady at the begining of the question |
| 11:49.59 | Dovid | ;) |
| 11:50.05 | shay|work | Dovid: I didn't ordered the card, we just got it from someone that does not needed it |
| 11:50.10 | shay|work | Dovid: yes, but shady is not me :P |
| 11:50.19 | Dovid | oops |
| 11:50.21 | Dovid | shay* |
| 11:50.28 | Dovid | so u can just order the daughter cards |
| 11:51.20 | shay|work | Dovid: great, that sounds like a cool option |
| 11:51.28 | shay|work | Dovid: I'll try to call Atelis |
| 11:51.50 | shay|work | dinner time, be back later |
| 11:51.51 | Dovid | i believe u can get it from digium direct |
| 11:51.54 | Dovid | later |
| 11:51.59 | shay|work | Dovid: thanks for your help |
| 11:57.05 | Dovid | np |
| 12:02.13 | ligoban | Dovid: the question is: I installed Asterisk to /misc/ast: ./configure --prefix /misc/ast --sysconfdir /misc/ast/etc --localstatedir /misc/ast/var |
| 12:02.28 | ligoban | Dovid: When I try to install asterisk-addons 1.4.0 (while doing make) |
| 12:02.41 | ligoban | >./configure --prefix /misc/ast --sysconfdir /misc/ast/etc --localstatedir /misc/ast/var --without-mysqlclient |
| 12:02.42 | ligoban | >make |
| 12:02.52 | ligoban | I get the message: |
| 12:02.52 | ligoban | <PROTECTED> |
| 12:02.52 | ligoban | configure: error: No asterisk installation found |
| 12:03.05 | Dovid | dont know 1.4 sorry |
| 12:03.14 | Dovid | y dont u install it to defualt DIR ? |
| 12:03.49 | ligoban | Dovid: defualt is not acceptable in my case |
| 12:04.06 | Dovid | i never used the configure |
| 12:04.09 | ligoban | Dovid: default installation is ok |
| 12:04.11 | Dovid | i am sticking with 1.2 |
| 12:04.23 | Dovid | i wouldt know how to point the add-ins |
| 12:04.24 | Dovid | ond* |
| 12:04.28 | Dovid | ons* |
| 12:04.30 | ligoban | Dovid: can you please advise me someone who knows? |
| 12:04.31 | Dovid | argh... i am tooo tired |
| 12:04.35 | Dovid | TK |
| 12:04.41 | Dovid | he should be on soon |
| 12:04.57 | ligoban | ok, I'll wait |
| 12:05.28 | Dovid | he comes on at diff times. |
| 12:05.30 | *** join/#asterisk af_ (n=getsmart@ip-156-32.sn2.eutelia.it) |
| 12:14.17 | shay | Dovid, I talked to Atelis, they said that it's no problem to change the doughter card with them :) |
| 12:16.36 | Dovid | shay: good |
| 12:16.48 | Dovid | what r they charging ? shop around |
| 12:17.53 | Dovid | shay: get my PM ? |
| 12:23.43 | voltagex | need some testers for my voice menu |
| 12:25.51 | *** join/#asterisk Corydon76-home (i=one@pdpc/supporter/sustaining/Corydon76-home) |
| 12:25.51 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 12:26.56 | Dovid | voltagex: what do u need to be tested ? |
| 12:27.12 | voltagex | Dovid: just that it works for external users |
| 12:27.27 | voltagex | I've had some strange connection issues with my box |
| 12:28.38 | Dovid | is it behind NAT ? |
| 12:29.15 | voltagex | Dovid: it should all be sorted now, can I msg you the details? |
| 12:29.23 | Dovid | sure |
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| 12:46.48 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:46.55 | Dovid | ~book |
| 12:46.56 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 12:52.35 | ligoban | Does anyone know how to install *addons-1.4.0 if I've installed * completely not to the default directory (/opt/asterisk)? |
| 12:56.09 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 12:58.28 | Dovid | ligoban: try the users list |
| 13:00.09 | ligoban | Dovid: nobody knows. |
| 13:04.12 | *** join/#asterisk garreel (n=garreel@host48-2-dynamic.16-87-r.retail.telecomitalia.it) |
| 13:06.04 | garreel | any xperience on zap channels fax detection? |
| 13:07.11 | garreel | tried to originate a call to a fax machine but seems no way to jump to the "fax" extension |
| 13:08.01 | garreel | tried all faxdetect=both/incoming/outgoing on zapata.conf |
| 13:10.59 | garreel | any human here?... or only answering machines? ;-) |
| 13:12.03 | Dovid | hehe |
| 13:12.08 | Dovid | never worked with faxing |
| 13:12.17 | garreel | :-) |
| 13:12.29 | Dovid | if some one has an answer they will respond |
| 13:21.48 | tzafrir | garreel, you don't *have* to use the fax detection. If you originate a call that you know is for faxing, just send it to the proper destination |
| 13:22.15 | tzafrir | garreel, on this channel you'll normally find human answering machines |
| 13:22.19 | *** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
| 13:22.26 | garreel | :-) |
| 13:22.41 | tzafrir | and jbot |
| 13:23.22 | garreel | I need fax detection... becouse I don't know if on the answering channel there is a fax... and if it's a fax I need not to pass to my agents |
| 13:24.02 | garreel | I originate a call and I hope to find a human... |
| 13:24.20 | tzafrir | garreel, you can enable incoming or outgoing fax detection separately (or "both", or "none", of course) |
| 13:24.40 | garreel | but if I find a fax I need to reschedule that call some hours later |
| 13:25.16 | garreel | I tried all settings on zapata.conf (incoming or outgoing or both) |
| 13:27.31 | *** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
| 13:28.01 | garreel | but none seems to produce a jump to the fax extension |
| 13:28.26 | garreel | using * 1.4.2 |
| 13:29.00 | garreel | if I use AMD application the fax is detected as answering machine |
| 13:29.55 | garreel | do I need some 3rd party application such as NVFaxDetect for that purpose? |
| 13:31.40 | garreel | I think I shoudn't... elsewere why there is a 'fax' extension? |
| 13:33.04 | *** join/#asterisk SoftIce (n=phil@vc-196-207-45-253.3g.vodacom.co.za) |
| 13:33.09 | SoftIce | bonjour! |
| 13:33.12 | garreel | I was wondering... is there a difference between a real fax machine and a modem acting like a fax? |
| 13:33.36 | SoftIce | well, 1 is a fax machine 1 is a modem acting like a fax :) |
| 13:34.00 | garreel | in term of detection I mean ;-) |
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| 13:52.46 | *** join/#asterisk __freedom__lover (n=__freedo@201-43-199-65.dsl.telesp.net.br) |
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| 13:52.49 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 13:53.13 | __freedom__lover | hi all, good morning |
| 13:53.57 | *** join/#asterisk KpoH (n=AID@host-86-106-252-223.moldtelecom.md) |
| 13:54.03 | __freedom__lover | i have a simple doubt, a sip user can be in two contexts differents at same time? |
| 13:59.26 | *** join/#asterisk AbsTradELic (n=vldmr@201.79.156.219) |
| 13:59.41 | AbsTradELic | anyone here from digium ? |
| 14:00.15 | *** part/#asterisk __freedom__lover (n=__freedo@201-43-199-65.dsl.telesp.net.br) |
| 14:00.35 | *** join/#asterisk Fieldy (i=haEzhTIx@gentoo/contributor/Fieldy) |
| 14:08.43 | garreel | AbsTradELic: today only answering machines |
| 14:09.55 | blitzrage | if you have a question, you could just ask it... :) |
| 14:10.16 | JerJer | info@talkshoe.com |
| 14:10.21 | JerJer | er |
| 14:10.26 | JerJer | http://www.crazyillusion.com/ |
| 14:10.37 | JerJer | stupid clipboard |
| 14:11.09 | blitzrage | JerJer: I was kinda expecting something like that... :) |
| 14:11.41 | JerJer | yep - it worked on my parents and other family |
| 14:12.00 | JerJer | didja see this: http://www.google.com/tisp/ |
| 14:12.05 | JerJer | i love it |
| 14:13.35 | *** join/#asterisk shadou (i=aj@unaffiliated/dj-fu) |
| 14:14.01 | coppice | most people in the broadband industry already think revenues have gone down the toilet |
| 14:15.04 | JerJer | lol - i kinda wonder if that's google's offical position as well - on that topic |
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| 14:15.06 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 14:16.21 | JT | http://mail.google.com/mail/help/paper/ |
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| 14:44.59 | hphinc | need some help: registration script for Business Edition is causing a segmentation fault. How to fix? |
| 14:47.06 | tzafrir | contact support@digium.com ? |
| 14:47.25 | blitzrage | wouldn't that be the reason you bought a business edition? |
| 14:47.30 | tzafrir | (or use strance and such) |
| 14:47.40 | tzafrir | strace |
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| 15:21.44 | stony | how can i configure mISDN devices with the digium webfrontend ? |
| 15:21.59 | blitzrage | very carefully |
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| 15:26.36 | Fieldy | heh. |
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| 15:51.14 | piper69 | good morning guys |
| 15:52.34 | piper69 | i have lucent Technoliges traning matirals for a 5ESS DCS Switch, i would like to share it if someone wants to learn |
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| 16:17.35 | theblue | Hi all. |
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| 16:25.30 | bsd_tech | they call it net split sunday |
| 16:25.38 | joemon | yup |
| 16:25.39 | joemon | :) |
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| 16:27.47 | blitzrage | banana net split |
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| 16:29.34 | nasls_lsa | is there any oder package than mISDN for BeroNet ISDN cards ? |
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| 17:07.11 | Assid | heya |
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| 17:32.30 | IOscanner | I have setup a single T1 card with PRI services. Has anyone done this with 2 T1 cards. |
| 17:33.07 | *** join/#asterisk zoa (n=zoa@pirus.securax.be) |
| 17:33.07 | ManxPower | IOscanner: There is nothing different. |
| 17:33.10 | *** join/#asterisk samdip (n=sam@modemcable146.244-202-24.mc.videotron.ca) |
| 17:33.20 | IOscanner | Do I just pickup the next channel 26-50. |
| 17:33.47 | ManxPower | yup. |
| 17:33.48 | mocker | Woo, just got my first simple AGI script going. :) |
| 17:34.17 | IOscanner | okay an leave 25 for signaling same as 0. |
| 17:34.18 | ManxPower | if both cards use the same driver, you have to be careful, as the 2nd card might be considered channels 1-24 |
| 17:34.45 | ManxPower | IOscanner: huh? Channel 24 and 48 would be Dchan |
| 17:35.14 | ManxPower | 1-24 == first span, 25-38 would be the next span. |
| 17:35.32 | ManxPower | ...er 25-48 |
| 17:35.40 | ManxPower | I've only had 1 cup of coffee so far today |
| 17:36.01 | samdip | asterisk/linux noob - need help |
| 17:36.38 | ManxPower | samdip: what SPECIFIC question do you have? |
| 17:36.52 | IOscanner | sorry you are right |
| 17:36.55 | IOscanner | 25-48 |
| 17:37.05 | samdip | i have an spa3000 that i reset now I get a busy signal when I pickup |
| 17:37.10 | IOscanner | thanks |
| 17:37.24 | ManxPower | samdip: that is neither an asterisk question nor a linux question. |
| 17:37.50 | ManxPower | if you factory reset the SPA, you will need to configure it before you get dialtone |
| 17:37.57 | samdip | well I'm trying to register the spa3000 to my asterisk box |
| 17:38.49 | Vec | samdip : u prob get a busy signal because it is not registering, checkout astersikguru it has a very nice tut on setting up a spa3000 etc to register with asterisk |
| 17:38.51 | samdip | i left everything as default, and just added user name and password to match my sip.conf |
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| 17:39.34 | pillowhead | should i use the add-ons package 1.2.4 or 1.4.0 with Asterisk 1.2? |
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| 17:39.45 | ManxPower | samdip: I imagine that you would also have to add the server name to the SPA config. You need to be watching the asterisk console to any error messages. |
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| 17:39.50 | ManxPower | pillowhead: 1.2 |
| 17:39.56 | pillowhead | k thanks |
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| 17:40.19 | mocker | Is it still the case in Asterisk 1.4 that for AGI STDERR messages you need to watch the console? |
| 17:40.27 | *** join/#asterisk RoyK (n=roy@217-175-152.100710.adsl.tele2.no) |
| 17:40.28 | samdip | the console confirm not registering |
| 17:40.32 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 17:40.40 | RoyK | asterisk SPE sounds fun :) |
| 17:40.50 | samdip | I'm not sure which setting is wrong on my spa3000 |
| 17:40.50 | ManxPower | samdip: paste the message inidicating that is not registeriong |
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| 17:42.31 | samdip | <-- SIP read from 192.168.1.200:5060: |
| 17:42.32 | samdip | SIP/2.0 404 Not Found |
| 17:42.32 | samdip | To: <sip:192.168.1.200>;tag=7166af97d86cf0e1i0 |
| 17:42.33 | samdip | From: "asterisk" <sip:asterisk@192.168.1.33>;tag=as32cc6a1c |
| 17:42.33 | samdip | Call-ID: 577fc29b1d8a492b7f3f148e106fe51a@192.168.1.33 |
| 17:42.33 | samdip | CSeq: 102 OPTIONS |
| 17:42.34 | samdip | Via: SIP/2.0/UDP 192.168.1.33:5060;branch=z9hG4bK01a10a0c |
| 17:42.36 | samdip | Server: Sipura/SPA3000-3.1.7(GWg) |
| 17:42.38 | samdip | Content-Length: 0 |
| 17:42.54 | *** join/#asterisk brian (i=brian@unaffiliated/brian) [NETSPLIT VICTIM] |
| 17:43.07 | *** join/#asterisk juice (i=juice@mo-76-0-10-6.dhcp.embarqhsd.net) [NETSPLIT VICTIM] |
| 17:43.11 | oej | This has nothing to do with REGISTER requests |
| 17:43.59 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) [NETSPLIT VICTIM] |
| 17:44.27 | samdip | ok my web interface for spa3000 says not registered |
| 17:44.54 | *** part/#asterisk RoyK (n=roy@217-175-152.100710.adsl.tele2.no) |
| 17:46.27 | *** join/#asterisk bawb2 (n=bawb2@ip24-255-194-150.ks.ks.cox.net) |
| 17:47.15 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 17:47.15 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:49.50 | samdip | if i select no for: "make calls without register" - "recv calls without register" - "register" I get a dial tone |
| 17:50.32 | samdip | as soon as I try to register, it doesn't work |
| 17:51.25 | *** join/#asterisk Exhar (n=Roy@213-73-139-87.cable.quicknet.nl) |
| 17:55.05 | samdip | anybody have any insights? |
| 17:55.39 | *** join/#asterisk friedrich| (n=friedric@e177252093.adsl.alicedsl.de) |
| 17:57.48 | samdip | checking out asteriskguru.com |
| 18:09.11 | *** join/#asterisk ToyMan (n=Stuart@cpe-24-161-103-133.hvc.res.rr.com) |
| 18:09.40 | [TK]D-Fender | samdip, clearly your credentials don't match |
| 18:09.47 | *** join/#asterisk Cresl1n (n=matt@c-68-62-219-187.hsd1.al.comcast.net) |
| 18:09.47 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 18:11.07 | samdip | can it be some other setting in the spa configuration? |
| 18:12.37 | [TK]D-Fender | samdip, Doesn't matter who's side you adjust so ong as they mathc |
| 18:12.40 | [TK]D-Fender | match |
| 18:13.17 | *** join/#asterisk `Sauron (i=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 18:16.14 | samdip | I have 1234 as user name and "hello" password... my sip.conf only has 8 lines |
| 18:16.14 | samdip | I don't see the missmatch |
| 18:16.16 | zoa | its your username |
| 18:16.25 | zoa | no need to use username for a client |
| 18:16.30 | zoa | its the thing between the brackets that counts |
| 18:16.35 | zoa | [1234] |
| 18:17.07 | zoa | username is to register to other servers |
| 18:18.06 | samdip | username in my sip.conf? |
| 18:18.18 | *** join/#asterisk ctooley (n=ctooley@tx-71-2-127-90.dhcp.embarqhsd.net) |
| 18:18.20 | bsd_tech | morning I need help fixing the asterisk addons for 1.4 on bsd |
| 18:18.28 | bsd_tech | I am working to patch |
| 18:18.37 | bsd_tech | but the configure is not working |
| 18:18.48 | bsd_tech | iys not looking in /usr/local/include |
| 18:19.38 | samdip | zoa, do you mean my username in sip.conf? |
| 18:19.39 | *** join/#asterisk zeach (n=zeach@0x5552ce95.adsl.cybercity.dk) |
| 18:20.15 | mmartinn | samdip: He means the thing that looks like a context in sip.conf; like [1234]. |
| 18:21.07 | bsd_tech | you should look at the sip.conf that came with the install and learn it |
| 18:21.28 | bsd_tech | read the files before messing with them it helps you to learn to do things correctly |
| 18:21.46 | zoa | yes |
| 18:21.49 | bsd_tech | the have examples in it of extensioons |
| 18:22.10 | zoa | follow the example on asteriskguru, it should be mentioned there |
| 18:22.18 | zoa | or look at the sip.conf tutorial |
| 18:22.52 | bsd_tech | sorry to spout off I just get tired of of people wanting handholding who did not take time to learn from the existing files |
| 18:23.13 | samdip | yes my [1234] matches 1234 in spa web interface under user user ID and Auth ID |
| 18:24.12 | *** join/#asterisk n3tim (n=bugf0x@BHE200150040158.res-com.wayinternet.com.br) |
| 18:24.13 | samdip | I'm going through the asteriskguru tutorial now.. thanks guys |
| 18:24.19 | bsd_tech | in the spa you dont put {} |
| 18:24.21 | bsd_tech | [] |
| 18:24.32 | bsd_tech | you put just the number and the pasword |
| 18:24.58 | bsd_tech | in the sip .conf you have to have [] around the exten number to make it a context to match |
| 18:25.27 | bsd_tech | so when the device sends 4653 it matches the context in sip.conf og [4653] |
| 18:25.38 | bsd_tech | og/of |
| 18:26.22 | samdip | bsd_tech, I understand, what you mean, I hate to admit this but I bought asterisk for dummies, it was working at one point, but I messed up somewhere and I can't figure it out |
| 18:26.24 | bsd_tech | ok any devs here that can help with dev for a min |
| 18:26.30 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 18:26.56 | bsd_tech | thats why you back up files before chnaging them |
| 18:27.03 | *** join/#asterisk CruDs (n=crudi@ppp-70-247-203-146.dsl.snantx.swbell.net) |
| 18:27.46 | samdip | how do you backup spa3000 before it screws up? |
| 18:29.20 | mmartinn | Hey folks; on a PRI, how do you get the "new number" on a "this number has been changed" outcome? |
| 18:30.21 | bsd_tech | checking for /usr/include/asterisk/channel.h... no |
| 18:30.21 | bsd_tech | configure: error: No asterisk installation found |
| 18:30.22 | CruDs | Usually you call the telco |
| 18:30.27 | bsd_tech | take screen shots |
| 18:30.33 | *** join/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 18:31.04 | bsd_tech | you can take html screen shots of the pages |
| 18:31.36 | bsd_tech | I cant get the configure to work |
| 18:31.39 | samdip | will definitely do that once i get it to work... thanks guys |
| 18:31.43 | mmartinn | On analog lines, the telco plays the new number -- that information is totally unavailable with PRI? |
| 18:32.28 | *** join/#asterisk high-rez (i=gus@carrera.bourg.net) |
| 18:33.15 | high-rez | Does anyone know what version SLIN codec support was introduced in? |
| 18:33.19 | bsd_tech | I need help from a dev |
| 18:33.33 | CruDs | There is a delay when I dial my VM. It says "Extension" and then pauses. How can I get rid of that? It also says something like "comedian voicemail" or "canadian voicemail", how can I get rid of that? |
| 18:33.53 | *** join/#asterisk `p4r14h (n=j0sh@69.92.145.178) |
| 18:34.37 | mocker | samdip: There's an asterisk for dummies? |
| 18:34.57 | mocker | Wow. |
| 18:35.49 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-14.ph.ph.cox.net) |
| 18:36.08 | samdip | yes, i dont recommended, most of the configuration is the for the web interface version of asterisk... I prefer command line. |
| 18:36.23 | bsd_tech | the gui needs work |
| 18:36.39 | mocker | They use trixbox for it? |
| 18:36.52 | *** join/#asterisk brussel_ (n=brussel@cpe-24-165-7-252.san.res.rr.com) |
| 18:37.13 | samdip | AsteriskNow i believe |
| 18:37.17 | bsd_tech | freepbx is nice |
| 18:37.29 | bsd_tech | asteriskNow is the digium gui |
| 18:37.34 | mocker | Ahh. |
| 18:37.44 | mocker | I just saw the trixbox hardware appliance. |
| 18:37.48 | mocker | That looks pretty sweet. |
| 18:37.49 | bsd_tech | its all java script |
| 18:38.04 | bsd_tech | freepbx is a diff gui |
| 18:38.16 | bsd_tech | and then there is also thirdlane |
| 18:38.33 | samdip | did you guys have a background in telephony before you into asterisk? |
| 18:38.40 | bsd_tech | I did |
| 18:38.48 | bsd_tech | I worked for Nortel |
| 18:38.57 | bsd_tech | and Nortek Satcom |
| 18:39.11 | mocker | Sysadmin background for me. |
| 18:39.21 | zoa | not so much of a student for me |
| 18:39.23 | samdip | how long do you think a noob would take to pick it up? |
| 18:39.35 | bsd_tech | depends on the noob |
| 18:39.36 | zoa | to be able to use it, less than a week |
| 18:39.44 | bsd_tech | and the gui |
| 18:39.48 | zoa | to know a lot about it, about 6 months |
| 18:39.56 | bsd_tech | the asterisk gui needs work |
| 18:40.03 | samdip | ok, in that case how woud it take for a retard like me? |
| 18:40.17 | bsd_tech | if you take time to read and understand 2 months |
| 18:40.28 | zoa | read the books |
| 18:40.30 | zoa | the different websites |
| 18:40.36 | bsd_tech | but you have to pick a phone and stick with it |
| 18:40.37 | `p4r14h | samdip: what are you trying to do with asterisk? |
| 18:40.43 | zoa | and experiment experiment experiment |
| 18:40.49 | zoa | and you will do just fine |
| 18:41.22 | samdip | p4r14h right now, just make it work.... with spa3000 |
| 18:41.24 | bsd_tech | where are the devs |
| 18:41.50 | `p4r14h | have you tried AsteriskNOW? |
| 18:42.20 | samdip | ok you guys, will do more reading, experimenting, and will be back, hopefully with smarter questions. thank again for your help |
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| 18:44.03 | samdip | p4r, I haven't , but I'm determined to make the installation I have , to work |
| 18:46.25 | CruDs | how do you stop it from saying "comedian mail?" |
| 18:48.43 | cr4z3d | sudo gedit /etc/asterisk/extensions.conf |
| 18:48.55 | cr4z3d | oh wow.. this is not my terminal |
| 18:49.23 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 18:50.20 | CruDs | is there a way to records sounds via a phone? |
| 18:51.14 | zoa | yes its called record() |
| 18:51.39 | CruDs | so im guessing you have to set up an extension with record |
| 18:52.11 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
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| 18:52.19 | zoa | yes |
| 18:52.59 | bsd_tech | no |
| 18:53.33 | cr4z3d | you can do monitor() |
| 18:53.43 | bsd_tech | Curd you cant |
| 18:54.03 | bsd_tech | comedian mail is the name of the voicemail system |
| 18:54.25 | zoa | i think he is just asking for recordings in general |
| 18:55.01 | bsd_tech | crap |
| 18:55.11 | bsd_tech | my brain hurts |
| 18:55.14 | CruDs | it says "comedian mail.. extension (big pause)" |
| 18:55.23 | bsd_tech | ok wait |
| 18:55.34 | bsd_tech | ok |
| 18:56.59 | CruDs | i'd rather it just go straight into the mailbox |
| 18:57.37 | CruDs | anyone use the GBX-2000 Grandstream phones? |
| 18:57.38 | bsd_tech | xten => *85,1,Answer |
| 18:57.38 | bsd_tech | exten => *85,2,Wait(1) |
| 18:57.38 | bsd_tech | exten => *85,3,Macro(user-callerid) |
| 18:57.38 | bsd_tech | exten => *85,4,VoicemailMain(${CALLERIDNUM}@default) |
| 18:57.39 | bsd_tech | exten => *85,5,Macro(hangupcall) |
| 18:57.41 | bsd_tech | there |
| 18:57.58 | bsd_tech | but that needs the macro |
| 18:58.05 | Dovid | bsd_tech: where r u getting the macro's from ? |
| 18:58.25 | CruDs | im guessing that goes in extensions.conf? |
| 18:59.08 | [TK]D-Fender | Just strip them. Who the hell needs 'em. |
| 18:59.19 | *** join/#asterisk jordanb (n=jordanb@68-20-177-188.dsl.chcgil.ameritech.net) |
| 18:59.21 | bsd_tech | ;;voice mailman with exten/callerid checking. if callerid/exten match |
| 18:59.21 | bsd_tech | ;prompts for password eslse prompts for mailbox/password |
| 18:59.21 | bsd_tech | exten => *85,1,VoicemailMain(${CALLERIDNUM}) |
| 18:59.21 | bsd_tech | exten => *85,n,Macro(hangupcall) |
| 18:59.25 | bsd_tech | I write them |
| 18:59.38 | jordanb | Will the vonage suit affect asterisk? |
| 18:59.54 | bsd_tech | asterisk does not work with vonage |
| 18:59.59 | bsd_tech | last I knew |
| 19:00.18 | bsd_tech | ddi they open it up to support asteriks now |
| 19:00.27 | jordanb | Well, it was a patent suit about bridging PSTN and VOIP though. |
| 19:00.57 | CruDs | so you can't just make it go straight to the voicemail and not play the intro message? im talking about internal use such as checking your vm, not for when getting a call. |
| 19:01.14 | bsd_tech | yes you can I just showed you how |
| 19:01.22 | bsd_tech | <bsd_tech> exten => *85,1,VoicemailMain(${CALLERIDNUM}) |
| 19:01.22 | bsd_tech | <bsd_tech> exten => *85,n,Macro(hangupcall) |
| 19:01.35 | CruDs | do you need the macro for that |
| 19:01.53 | bsd_tech | <bsd_tech> exten => *85,n,Hangup |
| 19:02.01 | bsd_tech | rm the macro |
| 19:02.19 | cr4z3d | what exactly is a "marco" i never understood the concept of it |
| 19:02.30 | bsd_tech | when I finish writing the new dialplan I will post for all |
| 19:02.56 | bsd_tech | look |
| 19:03.18 | bsd_tech | if you dont know what your dialpplan is and macros you need to read the files and learn |
| 19:03.24 | zoa | it will not affect asterisk |
| 19:03.36 | zoa | but might affect digium if and if and if |
| 19:03.38 | cr4z3d | i know what a dialplan is i just don't get the point of marcos |
| 19:03.44 | bsd_tech | macros are sup calls in a dialplan |
| 19:03.45 | zoa | asterisk is open source, not much they can do about it |
| 19:03.51 | CruDs | bsd: that didnt work |
| 19:03.57 | bsd_tech | it allows functionality |
| 19:04.06 | bsd_tech | you have to restart asterisk |
| 19:04.13 | bsd_tech | for it to load the new dial plan |
| 19:04.14 | cr4z3d | ok so if i setup a marco and call it during a dialplan it'll basically do another dialplan? |
| 19:04.16 | CruDs | i did |
| 19:04.17 | zoa | a macro is like a template |
| 19:04.22 | zoa | meaning something to make your code smaller |
| 19:04.31 | zoa | for repetitions |
| 19:04.33 | cr4z3d | ok so something i use over and over again |
| 19:04.36 | zoa | yes |
| 19:04.37 | cr4z3d | kind of like a function? |
| 19:04.40 | bsd_tech | yes |
| 19:04.44 | zoa | yes |
| 19:04.45 | [TK]D-Fender | cr4z3d , CruDs READ THE BOOK |
| 19:04.46 | [TK]D-Fender | ~book |
| 19:04.51 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:05.08 | bsd_tech | thank you |
| 19:05.13 | CruDs | i did, thank you |
| 19:05.24 | [TK]D-Fender | CruDs, Go over that chapter again then :) |
| 19:05.53 | [TK]D-Fender | bsd_tech, And don't give them sample that are dependant on MORE funny macro's. Their solution was a single dial-plan line. |
| 19:06.56 | *** join/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
| 19:06.59 | [TK]D-Fender | bsd_tech, Compounding this is the fact your sample is using 2 generation old CallerId vars that were deprecated in 1.2 and removed ENTIRELY from 1.4 |
| 19:07.07 | cr4z3d | i think i have a pretty good grasp on the basics for nwo at least. i'm just waiting for my ATA and phones to come in |
| 19:07.26 | bsd_tech | I did not know he was on 1.4 |
| 19:07.38 | bsd_tech | I just grabed from my old dialplan |
| 19:07.52 | bsd_tech | I am working to update it for 1.4 now |
| 19:07.59 | CruDs | ah, thats why it didnt work |
| 19:08.08 | markit | hi, I've terminated a received IAX2 call, after some time I've seen in the console this warning: "ast_cdr_merge: CDR start disagreement for IAX2/82.155.555"... any clue? * 1.4.x svn |
| 19:09.17 | *** join/#asterisk l0rdr0ck (n=l0rdr0ck@75.31.197.229) |
| 19:11.43 | CruDs | How do you utilize the LINE2 on the gbx-2000? |
| 19:11.48 | CruDs | the manual doesnt say much |
| 19:12.02 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 19:13.02 | Dovid | CrudDs: What do u need help with ? |
| 19:13.41 | CruDs | I'd like it so that when my second POTS line rings it lights up line 2 |
| 19:14.09 | Dovid | like 2 = ? |
| 19:14.22 | Dovid | what r u using? an IP phone ? ata ? |
| 19:15.33 | CruDs | gbx-2000 |
| 19:15.39 | cr4z3d | what's the quickest way to check if i have OHCI or UHCI for my timer? |
| 19:15.57 | CruDs | whenever my 2nd line rings, it still shows LINE1 ringing and it doesnt show caller id. it says "new user" and "asterisk" |
| 19:16.07 | cr4z3d | oh wait looks like i have ohci_hcd |
| 19:16.23 | [TK]D-Fender | cr4z3d, What os are you using? |
| 19:16.38 | cr4z3d | ubuntu |
| 19:16.52 | cr4z3d | lsmod has ohci_hcd |
| 19:17.10 | cr4z3d | and no usb_uhci |
| 19:19.34 | cr4z3d | oh nice looks like on the wiki i can use ztdummy regardless if i have 2.6 kernel |
| 19:20.30 | CruDs | anyone do Asterisk consulting? ie. provide paid help? |
| 19:22.37 | zoa | CruDs: yes |
| 19:22.43 | CruDs | i need my system working the way we need by tomorrow and we're ready to pay for help. |
| 19:22.56 | CruDs | there are only a few lingering issues |
| 19:25.41 | CruDs | http://forums.digium.com/viewtopic.php?t=14729 |
| 19:26.13 | Dovid | CruDs; Is it an ATA ? |
| 19:27.47 | *** part/#asterisk markit (n=konversa@host119-245-static.72-81-b.business.telecomitalia.it) |
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| 19:50.13 | [[blah]asfd | what is the default value for autokill= in iax.conf if nothing is defined? |
| 19:56.32 | blitzrage | [[blah]asfd: static int autokill = 0; |
| 19:57.34 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
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| 20:07.44 | ManxPower | CruDs: I do Asterisk consulting. I do not do emergency must be done by tomorrow consulting. |
| 20:14.58 | *** part/#asterisk CruDs (n=crudi@ppp-70-247-203-146.dsl.snantx.swbell.net) |
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| 20:21.55 | GreyFoxx | Anyone runaccross RealTime() improperly parsing a column ? Either it's Realtime chopping off the data or I'm just doing something wrong : IT's as if everything in front of the ";" is chopped off. http://www.pastebin.ca/420033 is info on what I've got in my config and what's coming out |
| 20:23.14 | Dovid | greyFoxx: what issue r u having ? that its not putting the entire info in to the db ? |
| 20:23.40 | Dovid | nm |
| 20:23.59 | GreyFoxx | Dovid: It's not filling ${var_fullcontact} with the full value of "fullcontact" from the database |
| 20:24.06 | GreyFoxx | Only the bit after a ";" |
| 20:24.39 | *** join/#asterisk `p4r14h (n=j0sh@69.92.145.178) |
| 20:25.31 | Dovid | i dont think user field is the same as user contact |
| 20:25.48 | Dovid | hang on. let em look at my db |
| 20:26.14 | GreyFoxx | userfield is just me filling in data in the CDR record. My problem is that Realtime is not returning the full "fullcontact" field from the sip users table |
| 20:26.39 | GreyFoxx | So either I'm not use it properly, or it's doing some wacky parsing and chopping off fullcontact before the ";" |
| 20:27.27 | GreyFoxx | Ok, I manually removed the ";" from the fullcontact table and now I get all of the data from that column leading me to believe that Realtime() is parsing it on the ";" |
| 20:27.43 | Dovid | what do u mean by ; ? |
| 20:28.04 | GreyFoxx | fullcontact contains " sip:2204@192.168.0.189:34540;rinstance=ac57c5859fecb786 " |
| 20:28.09 | Dovid | did u try nooping it ? |
| 20:28.16 | GreyFoxx | but realtime is only returning rinstance=ac57c5859fecb786 |
| 20:28.23 | *** join/#asterisk quizzy (n=walt@216.179.31.170) |
| 20:28.25 | GreyFoxx | if I remove the ";" I get all of it |
| 20:28.46 | Dovid | prob. cause mysql gets ; and thinks its then of the sql file |
| 20:28.53 | Dovid | sql command* |
| 20:28.54 | Dovid | sorry |
| 20:29.09 | GreyFoxx | Nah, the ; is in the data returned, not in the query being sent |
| 20:29.23 | Dovid | so all ur missing is hte port number ? |
| 20:29.30 | Dovid | the* from the CDR ? |
| 20:29.56 | GreyFoxx | No, I want the full fullcontact field, which includes the "sip:2204@192.168.0.189:34540 |
| 20:29.59 | GreyFoxx | portion |
| 20:30.03 | Dovid | if u do |
| 20:30.23 | Dovid | NoOp(${var_fullcontact}) what do u get ? |
| 20:31.28 | GreyFoxx | <PROTECTED> |
| 20:31.43 | GreyFoxx | It's chopping everything before the ";" in the returned column data |
| 20:31.58 | GreyFoxx | if I remove the ";" or exchange it for another character I get the entire thing |
| 20:32.09 | Dovid | so y dont u do that ? |
| 20:32.15 | Dovid | (sorry i am a bit tired here ) |
| 20:32.27 | GreyFoxx | Dovid: Cause I'm not the one populating fullcontact. Asterisk is :) |
| 20:32.41 | GreyFoxx | everytime someone registers |
| 20:33.20 | Dovid | where r u taking out the ; now that it is working ? |
| 20:33.24 | *** join/#asterisk zeeesh (n=aadilism@202.125.143.70) |
| 20:33.28 | zeeesh | hi |
| 20:33.46 | Dovid | hello zeeesh |
| 20:33.55 | GreyFoxx | Dovid: Manually in mysql I'm removing it , just to test my theory that Realtime() is chopping at the ";" |
| 20:34.35 | Dovid | it can be a bug |
| 20:34.44 | Dovid | email the users list |
| 20:35.20 | GreyFoxx | Sounds like I'll have too |
| 20:35.31 | Dovid | u may need a diffrent variable |
| 20:35.49 | Dovid | cause full contact is getting: sip:2204@192.168.0.189:34540;rinstance=ac57c5859fecb786 |
| 20:35.53 | zeeesh | i have been configure voicemail at my server ... i need to know .. is it possible to receive a copy of voicemail at your hotmail or yahooo or gmail or any webmail address??? |
| 20:36.12 | Dovid | userfiled is getting is coming up with the phone info as well |
| 20:36.12 | GreyFoxx | zeeesh: Just set the email address field for that account |
| 20:36.19 | Dovid | zeeesh: yes |
| 20:36.23 | Dovid | look in the wiki |
| 20:36.43 | Dovid | zeeesh: using real time or static ? |
| 20:37.14 | Dovid | http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf |
| 20:37.34 | zeeesh | <Dovid>:so wave file can also receive at your web address ? ok thanks |
| 20:37.37 | Dovid | there are examples there |
| 20:37.45 | zeeesh | ok thnx |
| 20:37.49 | Dovid | correct. to ur emai. u can also have a message sent to ur cell phone |
| 20:38.38 | zeeesh | <Dovid>:msg will free of cost or not ???? |
| 20:38.52 | Dovid | zeeeseh: to where ? ur email |
| 20:39.07 | Dovid | if sending a message to ur phone or pager it depends on ur carrier |
| 20:39.08 | zeeesh | cell phone |
| 20:39.16 | zeeesh | ok |
| 20:39.22 | Dovid | zeesh: where r u located ? |
| 20:40.07 | zeeesh | i m living in pakistan .. u can send sms through websites all the cellelur networks |
| 20:40.12 | Dovid | in the US cell providers give u an email address to send the message. if ur out of the US u may be SOL |
| 20:40.18 | joemon | dovid you use polycoms? |
| 20:40.24 | Dovid | used them in the past |
| 20:41.11 | Dovid | zeeesh: u have to ask ur provider if they have an email address that u can send a message to that will send it to the phone. the system its self cant send an sms (unless u have an sms gatway - even then i am unsure) |
| 20:41.13 | joemon | having issues with mine contacting vsftp server. Never recall this being a issue in the past |
| 20:41.19 | Dovid | joemon: they are my fav. phone |
| 20:41.28 | Dovid | what issues r u having ? |
| 20:41.29 | joemon | thay are nice thats for sure. |
| 20:41.32 | Dovid | when did it start ? |
| 20:41.33 | Dovid | lol. |
| 20:41.51 | zeeesh | <Dovid>: ok thnks .. |
| 20:41.54 | Dovid | np |
| 20:42.27 | joemon | well I dont recall vsftpd being a pain to configure regarding the upload of files. everything is set but when uploading the file get a 553 on ftp client. :) I need to do some more reading on this. |
| 20:42.35 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkit.com.au) |
| 20:44.09 | Dovid | hmm |
| 20:44.23 | Dovid | u have all the user names and passwords set that the phone is using ? |
| 20:44.43 | Dovid | when i use vsftpd i just start the service. create a user on the box and set that user on the phone and it gets it |
| 20:47.47 | joemon | yea seems that simple :) |
| 20:47.59 | Dovid | thats what i did and worked for me |
| 20:48.02 | joemon | ohh im beond that |
| 20:48.31 | Dovid | i would check the user id's and passworrs that r set on the phones. try localy on the box and see if u can get in |
| 20:48.42 | joemon | initially I could not upload files into the directory and got a 553 error in my client but found another way and just moved the files over. |
| 20:48.56 | joemon | yea thats been done with the phone. |
| 20:49.11 | Dovid | upload from the phone to the server ? |
| 20:49.12 | joemon | wow |
| 20:49.32 | Dovid | afaik the only files that get sent r the log files. |
| 20:49.41 | joemon | got a image erorr on my phone and it crashed. Well its obvios it is uploading the image but dont recall ever seeing this issue. |
| 20:50.02 | Dovid | i didnt know that u can upload the image from the phone to the server |
| 20:50.10 | joemon | so now its rebooting on its own again. |
| 20:50.12 | Dovid | or the server to the phone ? |
| 20:50.17 | Dovid | yea. fun fun |
| 20:50.20 | joemon | no from server to phone |
| 20:50.40 | Dovid | ah ok. |
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| 20:54.42 | *** part/#asterisk bsd_tech (n=bsdtech@ppp-69-238-48-150.dsl.irvnca.pacbell.net) |
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| 20:59.42 | *** join/#asterisk ZX81 (n=ZX81@60-234-238-188.bitstream.orcon.net.nz) |
| 21:00.27 | *** join/#asterisk Dovid[Laptop] (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 21:01.07 | ZX81 | hey all, this may seem like a stupid question, but I was just checking my routes and noticed I don't have one from any of the 10 providers I use for USA cellphones. Are these calls just charged at long distance rates? |
| 21:01.35 | ZX81 | i.e. 0.009 P/M? |
| 21:01.53 | *** join/#asterisk quizzy (n=walt@216.179.31.170) |
| 21:02.47 | Dovid[Laptop] | for US cell phones ? |
| 21:02.51 | ZX81 | yeah |
| 21:02.56 | Dovid[Laptop] | they are charged the same as any land line |
| 21:03.01 | ZX81 | sweeeeeet |
| 21:03.05 | Dovid[Laptop] | what do u mean by no route ? |
| 21:03.18 | ZX81 | I only have usa proper |
| 21:03.21 | Dovid[Laptop] | in the US cell phones use the same area codes as landline |
| 21:03.24 | ZX81 | every other coutry has proper and cell |
| 21:03.27 | ZX81 | yeah sweet |
| 21:03.29 | Dovid[Laptop] | it should be included in US proper |
| 21:03.31 | ZX81 | figured as much |
| 21:03.33 | ZX81 | kewl |
| 21:03.35 | Dovid[Laptop] | in the US the cell user gets billed |
| 21:03.44 | ZX81 | most countries are ~15-25c |
| 21:03.53 | Dovid[Laptop] | yea |
| 21:03.54 | ZX81 | weird I never noticed it before |
| 21:04.03 | Dovid[Laptop] | now u know ;) |
| 21:04.29 | Dovid[Laptop] | joemon: u in here ? |
| 21:05.32 | *** join/#asterisk Plecebo (n=larry@D-128-208-60-94.dhcp4.washington.edu) |
| 21:12.47 | tld2 | I'm curious. If we look away from signaling, and just at media, how do you all use Asterisk? IVR? Voicemail? Transcoding? Proxy for nat traversal? |
| 21:14.10 | blitzrage | tld2: yes. |
| 21:15.03 | tld2 | blitzrage: Kinda curious about which of the above. |
| 21:15.05 | *** part/#asterisk zeach (n=zeach@0x5552ce95.adsl.cybercity.dk) |
| 21:15.25 | tld2 | And if you're doing everything in software, or if you're using Asterisk with a physical interface as well, such as a digium board, and a landline or T1 |
| 21:15.33 | Dovid[laptop] | i use it for everything |
| 21:18.12 | quizzy | I use it as a substitute for a telco central office switch even though it doesn't have every feature of a switch. We sell telephone service as a replacement for POTS and asterisk does the job. |
| 21:18.38 | *** join/#asterisk kgx0 (n=kgx@60.234.20.178) |
| 21:18.48 | blitzrage | tld2: I run a distributed vPBX system for an ITSP. I use OpenSER as the registration point which distributes calls to multiple Asterisk boxes for call processing |
| 21:19.10 | blitzrage | I don't use hardware interfaces -- all SIP / IAX2 (although we do have a single PRI for a customer) |
| 21:19.51 | tld2 | blitzrage: And you don't use digium boards for timing etc? |
| 21:20.07 | blitzrage | tld2: no, that's what a 2.6 kernel is for |
| 21:20.09 | quizzy | Same here - no hardware interface. Cisco routers than then PRI lines to a tandem. |
| 21:20.31 | tld2 | sounds good |
| 21:21.06 | blitzrage | although when I start to scale up I'm sure I'll start installing the G.729 transcoder board into my machines |
| 21:21.16 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 21:21.33 | tld2 | makes sense I guess. |
| 21:21.45 | tld2 | G.729 requires a license, right? |
| 21:21.48 | blitzrage | of course |
| 21:21.59 | blitzrage | but the transcoder boards take the transcoding off the CPU |
| 21:22.44 | tld2 | Isn't G.729 fairly lightweight? I mean, shouldn't you be able to handle quite some channels on something like a modern dualcore? |
| 21:23.12 | tld2 | (just curious about cost/performance of adding more transcoder machines with dualcores, rather than transcoder hardware in one machine) |
| 21:23.45 | blitzrage | transcoding is about the hardest thing you can make the CPU do |
| 21:25.27 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 21:26.01 | joemon | i need sip.ld for my ip500. |
| 21:27.37 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 21:28.28 | joemon | k |
| 21:28.29 | sevard | blitzrage: besides frying eggs?! |
| 21:28.38 | blitzrage | indeed :) |
| 21:32.23 | *** join/#asterisk p4r14h (n=j0sh@69.92.145.178) |
| 21:35.34 | *** join/#asterisk svenna_ (n=svenna@p548d30e7.dip0.t-ipconnect.de) |
| 21:41.07 | Hmmhesays | haha googles april fools joke today is great |
| 21:41.52 | garreel | where is? |
| 21:42.58 | Hmmhesays | http://www.google.com/tisp/ |
| 21:43.05 | tld2 | "Why is TiSP in beta? |
| 21:43.06 | tld2 | When things go wrong with TiSP, they go very, very wrong. Let's leave it at that. |
| 21:43.47 | Hmmhesays | lol |
| 21:44.34 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 21:46.24 | sevard | Hmmhesays: the gmail one is better |
| 21:52.34 | *** join/#asterisk Mavvie (n=edwin@ppp245-76.lns3.syd6.internode.on.net) |
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| 22:02.15 | *** part/#asterisk [[blah]asfd (n=ckwall@c-71-195-199-149.hsd1.ut.comcast.net) |
| 22:03.30 | JT | gmail paper |
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| 22:06.29 | sevard | OUR BEES WILL BLOT OUT THE SUN |
| 22:06.34 | sevard | THEN WE WILL HUG IN THE SHADE. |
| 22:06.41 | *** part/#asterisk garreel (n=garreel@host48-2-dynamic.16-87-r.retail.telecomitalia.it) |
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| 22:25.08 | Vec | How can I check that speex has been compile correctly into asterisk, without actually testing it ;) ? |
| 22:25.15 | Vec | show codecs or something ? |
| 22:25.51 | *** join/#asterisk voltagex (n=voltagex@124-254-124-5-dsl.ispone.net.au) |
| 22:28.30 | *** join/#asterisk JunK-Y (n=junky@modemcable140.185-70-69.mc.videotron.ca) |
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| 22:40.27 | GreyFoxx | Well it's definately code splitting on the ;, now to find our why or if I can get around it :) |
| 22:41.41 | russellb | GreyFoxx: hm? |
| 22:42.14 | GreyFoxx | using Realtime to retrieve a value from a column in the database where the data contains ";" |
| 22:42.28 | GreyFoxx | IT's not returning the whole value, but just the part after the ; |
| 22:42.39 | russellb | oh, interesting ... |
| 22:42.55 | russellb | have you tried escaping it in the DB? ... whatever\;something ... |
| 22:42.55 | GreyFoxx | it's intentionally splitting the value on the ;, I just don't see why at this point |
| 22:43.10 | GreyFoxx | russellb: It's not something I'm putting in the database, asterisk is |
| 22:43.18 | GreyFoxx | so without code mods I can't change that |
| 22:45.36 | piper69 | hi all |
| 22:45.36 | GreyFoxx | More than anything I'm trying to figure out just why it's splitting the data instead of just returning it |
| 22:45.57 | piper69 | i just installed asterisk, how can i run it pleae |
| 22:46.02 | piper69 | *please |
| 22:46.59 | blitzrage | ~docs |
| 22:47.08 | jbot | docs is, like, Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 22:48.17 | russellb | believe it or not, you can run it by typing "asterisk" |
| 22:50.29 | piper69 | russellb: i did that and nothing happened , i know i have to read and read alot, i just wanted a jump startg |
| 22:50.29 | piper69 | :) |
| 22:50.29 | russellb | well, it runs in the background by default |
| 22:50.45 | piper69 | russellb: please, i just have few questions |
| 22:50.49 | russellb | asterisk -vvvvgc |
| 22:51.19 | piper69 | 1) do i need an ATA to run it or i can config my headset |
| 22:51.32 | piper69 | 2) how to config the headset |
| 22:51.45 | *** join/#asterisk krapper (n=krapper@wsip-64-58-154-130.oc.oc.cox.net) |
| 22:51.47 | piper69 | 3) how to look for a voip provider |
| 22:52.18 | russellb | 1) no. 2) look up chan_alsa. 3) google. :) |
| 22:52.54 | piper69 | russellb: thats some dry shit man |
| 22:52.55 | piper69 | hahahhaha |
| 22:53.06 | JT | it's a good summary |
| 22:53.09 | russellb | sorry, i'm about to leave :-p |
| 22:53.18 | piper69 | JT: i bet huh |
| 22:53.41 | piper69 | anyone here know what is a 5ESS DCS? |
| 22:54.22 | GreyFoxx | google does |
| 22:54.41 | piper69 | GreyFoxx: do you know? |
| 22:55.09 | GreyFoxx | Only what google told me in the summaries of the results when I searched for it :) |
| 22:55.36 | piper69 | GreyFoxx: so you don't know ? |
| 22:56.09 | GreyFoxx | I assume english isn't your first language ? |
| 22:56.33 | GreyFoxx | No, I do not. I know as much as I learned in 3 seconds looking at google search results, which is pretty much nothing |
| 22:56.54 | piper69 | i am a 5ESS DCS switch engineer |
| 22:57.26 | piper69 | my point is that no one knows everything from the day the are born, they learn , |
| 22:57.32 | Makenshi | piper69, what exactly do you need help with? |
| 22:57.39 | krapper | using an internal SIP client, you dial out to the PSTN (Zap), you talk for 5 minutes, then the call is transfered to another SIP client and they talk for 10 minutes (same asterisk server) then hangup. The CDR will show two entries, first for 5 minute call (SIP > Zap - BILLABLE) The second line shows from Zap > SIP for 10 minutes, which looks the same as an inbound call from PSTN in the CDR. (Typically NON-BILLABLE) Any ideas? |
| 22:58.00 | GreyFoxx | piper69: Absolutely, which is why so many have written manuals and documentation on so many subjects throughout history :) |
| 22:58.10 | piper69 | and i just wanted to learn asterisk and wanted a guide |
| 22:58.30 | blitzrage | piper69: www.asteriskdocs.org |
| 22:58.35 | piper69 | and since you guys used it for a while i wanted to come here for you advice |
| 22:58.37 | GreyFoxx | One of the best resources you will find is voip-info.org I can usually find something there, or asteriskdocs.org |
| 22:58.56 | blitzrage | piper69: your best bet is to ask a specific question -- this isn't really a room for hand holding |
| 22:59.17 | piper69 | blitzrage: i figuer this out ;) |
| 22:59.54 | blitzrage | if we held the hand of every person in here, some people would never sleep or go to work |
| 22:59.54 | piper69 | btw if any is intrested in learning the lucent 5ESS i have a good selection of cd's |
| 23:00.36 | piper69 | blitzrage: you can start by holding my hand, then we will see if you tired yoiu can go lolololol |
| 23:00.39 | piper69 | jk |
| 23:00.55 | Makenshi | piper69, sure, if you wouldn't mind sharing :) |
| 23:00.56 | blitzrage | piper69: I wrote a book... I paid my dues :) |
| 23:01.07 | piper69 | cool , i guess i have some reading to do |
| 23:01.14 | blitzrage | yep, that's how you learn |
| 23:01.23 | blitzrage | or you're just going to ask questions that have already been answered in documentation |
| 23:01.26 | GreyFoxx | and break things |
| 23:01.37 | blitzrage | I wrote it so I wouldn't have to answer the same questions over and over |
| 23:01.38 | GreyFoxx | you learn more by just trying (and failing) and fixing than anything |
| 23:01.44 | blitzrage | it's true |
| 23:02.01 | piper69 | i belive so too |
| 23:02.20 | blitzrage | good -- then we're all in agreement |
| 23:02.22 | blitzrage | !tfot |
| 23:02.24 | blitzrage | errr |
| 23:02.26 | blitzrage | ~tfot |
| 23:02.27 | jbot | i heard tfot is "The Future of Telephony", a book about Asterisk from O'Reilly Publishing, ISBN: 0-596-00962-3, click http://www.oreilly.com/catalog/asterisk/ for more details |
| 23:02.29 | piper69 | but for somereason the books the talk talk talk |
| 23:02.47 | JT | what? |
| 23:03.06 | piper69 | sorry |
| 23:03.18 | blitzrage | oooo..... 7pm.... nearly hockey time! |
| 23:03.32 | HockeyInJune | :) |
| 23:03.37 | blitzrage | Raptors are already winning! |
| 23:03.40 | piper69 | i meant to say some books they just keep talking about things not related |
| 23:04.08 | JT | the only book we tend to recommend is the O'Riely A:TFOT one |
| 23:04.21 | JT | O'Reilly even |
| 23:04.34 | *** join/#asterisk mroli (n=mroli@wsip-64-58-154-130.oc.oc.cox.net) |
| 23:04.36 | mvanbaak | orly |
| 23:04.45 | blitzrage | Asterisk is a cornucopia of 125 years of telephony packaged into one peice of software -- there is going to be a TON of things to know |
| 23:05.10 | blitzrage | if it's telephony related -- it is Asterisk related |
| 23:05.11 | mroli | hello all |
| 23:05.12 | mroli | anybody familiar with CDR and billing ? |
| 23:05.18 | blitzrage | what's your question? |
| 23:05.24 | *** join/#asterisk BosHaus (i=nobody@supporter/active/BosHaus) |
| 23:05.37 | mroli | having a hard time with transferred calls... |
| 23:05.43 | mroli | they show up as 2 different lines |
| 23:05.48 | BosHaus | anyone know any good guides on setting up asterisk (using my current phone line) |
| 23:05.50 | mroli | on the cdr... and trying to find a way to "group" them |
| 23:05.59 | BosHaus | all the guides I find are for IAX/SIP termination and whatnot |
| 23:06.08 | BosHaus | i've been browing voip-info.org |
| 23:06.56 | JT | look up zaptel |
| 23:07.09 | Vec | I get an error "no translater path exists for channel type zap" when I try make a call from a IAX phone that support speex on a ZAP channel, is this because speex is not installed correctly or could it be related to something else ? |
| 23:07.29 | piper69 | JT: when you say look up zaptel, where should i look up |
| 23:07.32 | JT | Vec: most likely the first situation |
| 23:07.36 | mroli | anyone has ever tried to do that with the CDR ? |
| 23:07.48 | JT | piper69: i was saying BosHaus should look it up, but i guess it could be useful for you too |
| 23:08.04 | Vec | JT : is tehre a way I can check if speex is installed, by typing something on the cli ? |
| 23:08.07 | *** join/#asterisk `p4r14h (n=j0sh@69.92.145.178) |
| 23:08.11 | piper69 | JT: i know , but look up where? google? |
| 23:08.22 | JT | ~thewiki |
| 23:08.23 | jbot | from memory, thewiki is at http://www.voip-info.org/wiki-Asterisk |
| 23:08.25 | JT | or |
| 23:08.27 | JT | ~thebook |
| 23:08.29 | jbot | i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:08.44 | Vec | thats a great book |
| 23:09.16 | krapper | that's more than a great book... it's the BIBLE! |
| 23:10.51 | blitzrage | mroli: basically that's how the CDRs are going to work right now -- in the later versions of Asterisk, the CDR system is going to be overhauled (work and discussion has already started) -- for now, you might want to utilize the userfield column to somehow put some information in the records to relate them together |
| 23:11.06 | blitzrage | and a 2nd edition will be out soon enough :) |
| 23:11.46 | mroli | blitz: exactly what i tried to do... using userfield to tag the CDR before placing an outbound call... once the call is transferred, on the second entry of the CDR, the userfield becomes empty. |
| 23:12.13 | mroli | blitz: also tried to use the AMflag, and flag the call with "billing" when dialing outbound... the flag is also lost on the second line after transferring teh call... |
| 23:13.02 | blitzrage | mroli: yah, for now, I haven't tried grouping them myself, so I don't have an answer for you unfortunately |
| 23:14.52 | mroli | blitz: there are many billing packages for asterisk out there... i guess they found a way to work around that... |
| 23:15.25 | mroli | blitz: or i guess i have to disable call transfer... :( how do you do that, btw ? |
| 23:17.46 | *** join/#asterisk l0rdr0ck (n=l0rdr0ck@adsl-75-31-197-229.dsl.pltn13.sbcglobal.net) |
| 23:22.11 | BosHaus | I was trying to mostly follow the guide on systm (http://revision3.com/systm/asterisk) but they don't show the options to put into the sipura 3100 :\ |
| 23:23.06 | *** join/#asterisk AMUG (n=junky@modemcable140.185-70-69.mc.videotron.ca) |
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| 23:23.26 | hillct | Good evening all |
| 23:23.56 | JT | good morning |
| 23:24.14 | hillct | How do I specify different registration timeouts different VSP register directives? |
| 23:24.29 | hillct | under asterisk 1.2 |
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| 23:26.15 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 23:29.31 | *** join/#asterisk `p4r14h (n=j0sh@69.92.145.178) |
| 23:29.38 | mroli | all: is there a way to disable call transferring from one SIP phone to another ? |
| 23:31.48 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 23:53.53 | *** join/#asterisk Nivex (n=kjotte@user-0ce2kma.cable.mindspring.com) |