| 00:05.59 | mmlj4 | ricer |
| 00:13.28 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 00:16.22 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 00:20.47 | *** join/#asterisk Akiyuki (n=jimi@cpe-065-184-197-243.ec.res.rr.com) |
| 00:21.51 | slaney | Soviet Russian? |
| 00:21.52 | slaney | fale |
| 00:22.33 | guax | lol |
| 00:22.52 | guax | jbot, about |
| 00:22.53 | jbot | http://www.infobot.org/ - /msg infobot help for a list of commands - http://www.infobot.org/guide-0.43.x.html for a guide |
| 00:23.43 | Akiyuki | heh |
| 00:24.04 | Akiyuki | Man, the fedora package for asterisk is way behind. Looks like I will be installing from source. |
| 00:24.35 | guax | i always installs from source |
| 00:26.46 | slaney | that bothers me. I think I am going to figure out how to package it for a few distros and make packages. |
| 00:27.57 | f0ner00t | How is the best way to set up an dialplan to answer the phone and than trans to an ext? |
| 00:28.31 | Akiyuki | s/than/then |
| 00:44.27 | f0ner00t | What is wrong ? [globals] |
| 00:44.27 | f0ner00t | [general] |
| 00:44.27 | f0ner00t | autofallthrough=yes |
| 00:44.27 | f0ner00t | [incoming] |
| 00:44.27 | f0ner00t | exten => 2537531745,1,Answer() |
| 00:44.30 | f0ner00t | exten => 2537531745,2,n,Background(main-menu) |
| 00:44.32 | f0ner00t | exten => 2537531745,3,n,WaitExten()exten => 1000,1,Answer() |
| 00:44.35 | f0ner00t | exten => 2537531745,4,n,Background(main-menu) |
| 00:44.37 | f0ner00t | exten => 2537531745,5,n,WaitExten() |
| 00:44.40 | f0ner00t | [1000] |
| 00:44.42 | f0ner00t | exten => 1000,1,Verbose(1|Extension 1000) |
| 00:44.45 | f0ner00t | exten => 1000,n,Dial(SIP/1000,30) |
| 00:44.47 | f0ner00t | exten => 1000,n,Hangup() |
| 00:44.50 | f0ner00t | opps Sorry. |
| 00:44.51 | slima | ~paste |
| 00:44.52 | jbot | i heard paste is http://rafb.net/paste/, or see also pb, or http://bin.cakephp.org/ |
| 00:44.52 | f0ner00t | That was suppose to be pastebin link. |
| 00:45.14 | Akiyuki | wtf |
| 00:45.23 | dwayne | you have 2 priorities |
| 00:46.44 | dwayne | and ... exten => 2537531745,3,n,WaitExten()exten => 1000,1,Answer() |
| 00:49.06 | f0ner00t | Ahh that should be 3 huh |
| 00:49.36 | f0ner00t | Or I don't need one. |
| 00:50.19 | f0ner00t | pbx_extension_helper: No application 'n,Background' for extension (incoming, 2537531745, 2) |
| 00:50.30 | f0ner00t | Does that mea IU need that application? |
| 00:50.35 | dwayne | no, you can have ',2,n', ',3,n', ... , and you are defining a new extension in the middle of your other one |
| 00:51.08 | dwayne | s/can/cant |
| 00:51.38 | f0ner00t | So one exten. |
| 00:52.56 | f0ner00t | Okay I have one ext. |
| 01:07.00 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 01:09.45 | f0ner00t | hmmp |
| 01:09.57 | f0ner00t | aNYBODY ROUND? |
| 01:10.30 | x86 | nO |
| 01:11.12 | f0ner00t | :) Cool. I see that. |
| 01:11.21 | troy- | anyone know how to create a NAPTR record for _sips._udp? |
| 01:12.31 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:16.00 | f0ner00t | Whats wrong w/ this statment ? see pastebin http://pastebin.com/m2fee3f07 |
| 01:18.51 | mmlj4 | line seven... do you have an extension 1? |
| 01:22.44 | f0ner00t | Which line is that mmlj4 ? |
| 01:23.18 | dwayne | f0ner00t, you may want to consider a GUI |
| 01:23.26 | AndyML | f0ner00t: looks like you might want to read the book too... |
| 01:23.31 | AndyML | ~book |
| 01:23.31 | jbot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 01:23.42 | f0ner00t | Andyml I read the book backwards and fowards. |
| 01:23.46 | f0ner00t | Gnno |
| 01:23.49 | f0ner00t | Hmmp. |
| 01:24.03 | f0ner00t | dwayne. You mean the ISO asterisk now? |
| 01:24.25 | AndyML | go back to the dialplan section - your dialplan is a good start but it looks like your understanding is a little shaky |
| 01:24.33 | dwayne | no, the AsteriskGUI is what I was thinking of |
| 01:25.01 | AndyML | f0ner00t: take a look. - http://pastebin.com/m38a24c30 |
| 01:25.10 | AndyML | see the difference? |
| 01:25.16 | mmlj4 | f0ner00t: you pasted it, I'm just reading what you posted |
| 01:25.25 | dwayne | f0ner00t, try typing an example EXACTLY as detailed in the book and you'll get the hang of it ... or look at the sample extensions.conf |
| 01:28.59 | f0ner00t | dwayne: I did that too.. |
| 01:29.10 | f0ner00t | AndyML: Thanks I will check it out./ |
| 01:29.43 | dwayne | if you did that, it would have worked |
| 01:29.49 | f0ner00t | AndyMl... Got it.. |
| 01:29.56 | f0ner00t | I understand where I was mistaken now. |
| 01:30.00 | f0ner00t | Okay gotta go.. beback later. |
| 01:30.02 | f0ner00t | Thank you |
| 01:30.08 | f0ner00t | I've been reading the book all day. |
| 01:30.23 | dwayne | yay |
| 01:44.52 | *** join/#asterisk saftsack (n=saftsack@p57924987.dip.t-dialin.net) |
| 01:47.23 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 01:57.01 | surfdue | hola |
| 01:57.51 | surfdue | :} |
| 01:59.05 | guax | sorry about the nick changing |
| 02:00.32 | surfdue | guax: you are so many peple ;} |
| 02:01.12 | guax | i have identity issues |
| 02:01.14 | guax | =P |
| 02:01.21 | surfdue | lol. |
| 02:01.31 | surfdue | are you an asterisk expert? |
| 02:01.39 | guax | no, just user |
| 02:01.41 | guax | =] |
| 02:01.42 | surfdue | :} |
| 02:01.51 | surfdue | i was ganna try freeswitch |
| 02:01.57 | surfdue | but i use to use asterisk |
| 02:02.06 | surfdue | i dont know either though |
| 02:02.10 | surfdue | anymore atlaest.. :[ |
| 02:19.04 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 02:19.16 | phix | hey |
| 02:19.43 | phix | any way that playtones can play the same tone that the caller is listening to when they dial? |
| 02:20.04 | phix | the different sound makes some people shit themselfs and hang up, I dont want that :) |
| 02:22.22 | *** join/#asterisk Badrobot- (i=Badrobot@cpe-76-173-233-75.socal.res.rr.com) |
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| 02:25.14 | maximo_ | hello phix: |
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| 02:40.04 | *** join/#asterisk jjg (n=jjg@76.21.4.40) |
| 02:40.21 | jjg | is a 404 not found error mean that asterisk could not find the user in the sip.conf? |
| 02:40.55 | SwK_ | jjg, it might mean that the extension was not found in the dialplan |
| 02:41.34 | jjg | Swk_ .. oh yah, makes sense .. i was trying to dial user@myasteriskserver ..but i should be dialing the extension, right? |
| 02:45.07 | SwK_ | jjg, if 'user' exists int eh dialplan you can call it... |
| 02:45.26 | SwK_ | extension => username,1,dial(SIP/user) works |
| 02:45.37 | SwK_ | where username is like for instance swk |
| 02:46.02 | SwK_ | everyone seems to want to think the extensions must be numberic but they dont have to be |
| 02:47.18 | jjg | SwK_ ..ok, got it. thanks, i am able to connect now |
| 02:47.27 | SwK_ | np |
| 02:55.26 | *** join/#asterisk docid (n=hagbard@69.196.68.142) |
| 02:56.02 | docid | is there any way i can use the ,r option in Dial command to signal ringing but still pass all the audio from the channel being connected to the caller? |
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| 03:47.18 | docid | is there any way i can use the ,r option in Dial command to signal ringing but still pass all the audio from the channel being connected to the caller? before that channel answers? |
| 03:51.01 | troy- | does anyone have examples of H323 SRV and NAPTR records? |
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| 04:48.58 | f0ner00t | `/quit |
| 04:49.09 | troy- | fail. |
| 04:49.33 | surfdue | LOL |
| 04:49.43 | surfdue | <PROTECTED> |
| 04:49.56 | troy- | /woo is way cooler |
| 04:50.21 | surfdue | haha |
| 04:50.28 | surfdue | <PROTECTED> |
| 04:50.30 | surfdue | jk |
| 04:50.30 | surfdue | :} |
| 04:50.41 | troy- | if i could, i would |
| 04:50.49 | surfdue | if you would, you could ;] |
| 04:51.30 | troy- | touche. |
| 04:52.03 | surfdue | toosay, |
| 04:57.17 | [TK]D-Fender | docid: No |
| 04:58.03 | docid | ok, no way to get the effect, or just no way to do it with the Dial command? |
| 04:58.59 | [TK]D-Fender | Dovid: You aren't getting BOTH. "r" does what it does, without it you know what you get. There is no "combined" |
| 05:00.08 | docid | ahh, hrmm, since we switch the link between the iMG and the asterisk box to pri, and the telco line is R1 MF, we seem to have lost the passthrough of ringing signals |
| 05:00.15 | docid | thats why im looking |
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| 05:16.43 | drmessano | yawn |
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| 05:45.56 | *** mode/#asterisk [+o mog] by ChanServ |
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| 06:29.00 | lucidblue | hey guys, I'd like to have a caller go to queue2 if they've been holding for more than 3 minutes... what would the line be like for something like that... what app would I use? |
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| 07:22.39 | jblack | lucidblue: Timeout the wait for the first queue, then dial the second queue |
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| 08:15.22 | aniasis | Hello |
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| 09:33.16 | aniasis | hey |
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| 11:56.23 | aniasis | Hello |
| 11:56.56 | tzafrir_laptop | ~ask |
| 11:56.56 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 11:57.20 | tzafrir_laptop | is here against our will |
| 11:58.57 | mvanbaak | too |
| 11:59.18 | tzafrir_laptop | aniasis, wanted to ask anything? |
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| 12:08.07 | *** join/#asterisk trask (n=trask@1407ds1-bav.0.fullrate.dk) |
| 12:08.25 | trask | hello |
| 12:09.08 | trask | so if i dream of starting my personal garage phonecompany ive come to the right place? |
| 12:09.21 | mvanbaak | :) |
| 12:09.26 | mvanbaak | maybe |
| 12:10.10 | Daejeo | trask: where do you want to do phone business? |
| 12:10.30 | trask | copenhagen denmark, nothing to do with money, just for me and freinds |
| 12:10.44 | trask | get to control my own phone numbers ect |
| 12:11.41 | trask | i could move to russia or something, would that help? |
| 12:12.26 | Daejeo | do you want to receive calls in Russia from >copenhagen? |
| 12:13.09 | trask | just want a server that spits calls for my numbers to my ekiga clients |
| 12:15.39 | trask | and sudo apt-get install asterisk worked fine, figure next step would be an answer on wether i was facing tons of expenses or there was loopholes making that kinda stuff actually possible |
| 12:34.52 | *** join/#asterisk ludan (n=daniele@unaffiliated/ludan) |
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| 12:56.36 | *** join/#asterisk ph8 (i=ph8@unaffiliated/ph8) |
| 12:56.52 | ph8 | hi everyone - is there a way for me to listen to any messages without using a phone? (e.g. from a shell) |
| 12:57.25 | Akiyuki | sure, if the call has been recorded, you can play it back through the shell. |
| 13:05.23 | ph8 | that's great, is it just a wav file somewhere? |
| 13:07.13 | tzafrir_laptop | ph8, /var/spool/asterisk/voicemail/ |
| 13:07.59 | tzafrir_laptop | The package sox includes the command play |
| 13:09.25 | Akiyuki | yeah, sorry, i was in the shower |
| 13:12.39 | ph8 | thanks that's got it sorted, i should figure out how to make it email me when there's a voicemail |
| 13:18.08 | tzafrir_laptop | ph8, what distro is it? |
| 13:18.19 | ph8 | ubuntu |
| 13:18.36 | tzafrir_laptop | do you have postfix installed? |
| 13:18.41 | ph8 | no |
| 13:18.43 | ph8 | i should probably do that :) |
| 13:19.04 | slaney | AwayML: wake up |
| 13:20.18 | ph8 | tzafrir_laptop: Does Asterisk just send an email to the address associated with the voice mailbox now that postfix is installed? |
| 13:20.29 | tzafrir_laptop | ph8, yes |
| 13:20.44 | ph8 | thx :) |
| 13:21.07 | tzafrir_laptop | yes can also try: echo whatever | mail -s "the subject" username@example.org |
| 13:22.17 | ph8 | great, cheers |
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| 14:14.24 | Daejeo | hello slaney |
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| 14:20.08 | slaney | hello |
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| 14:28.44 | Dovid | slane: Hi there |
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| 14:32.06 | Daejeo | do you know Andy? |
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| 14:35.40 | Magicblaze007 | how can I tell if I got a PAP2T or PAP2T-NA ?? The firmware is 3.1.15(LS). Is PAP2T = PAP2T-NA? |
| 14:37.11 | coppice | the only difference between the different PAP2T variants is the plug on the mains lead |
| 14:38.35 | Magicblaze007 | Where is "-NA" written? My box has PAP2T written on it. I bought it from Amazon which said they shipped PAP2T-NA. |
| 14:39.06 | coppice | it says on the carton somewhere |
| 14:39.22 | Magicblaze007 | which is the latest firmware for it ? |
| 14:40.09 | coppice | 5.2.something, I think. even units recently out of the factory come with ancient firmware. you need to reflash them |
| 14:40.18 | Magicblaze007 | indeed the carton says PAP2T-NA! |
| 14:41.12 | [TK]D-Fender | NA was meant to say "not provider locked" |
| 14:41.23 | coppice | NA == north america |
| 14:41.26 | Magicblaze007 | coppice: you are right. I have a unit that was working without any problems in the US, but is giving trouble in Europe. I am thinking of asking my europe client to upgrade the firmware. |
| 14:42.06 | Magicblaze007 | ooo...should I ask the client to buy a PAP2T-EU? is there a firmware for PAP2T-EU? |
| 14:42.37 | Daejeo | NA- not applicable for u |
| 14:42.41 | coppice | like I said, the units are identical. the code letter just tell you about the power supply in the box |
| 14:42.42 | Daejeo | send it to me |
| 14:42.46 | slaney | Daejeo: I do |
| 14:42.50 | slaney | we work together |
| 14:42.55 | Daejeo | ah |
| 14:43.06 | Magicblaze007 | coppice: So I'll just ask my client to upgrade the firmware. |
| 14:43.06 | Daejeo | got you |
| 14:43.25 | coppice | yep |
| 14:43.32 | Magicblaze007 | Thanks. |
| 14:44.03 | Magicblaze007 | Is this the place to download: http://www.linksysbycisco.com/US/en/support/PAP2T/download |
| 14:44.06 | coppice | this is only true for the PAP2T. The PAP2 seems to be quite different in the way the code the suffix |
| 14:44.34 | coppice | guess :-) Cisco have been moving everything around |
| 14:49.19 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 14:54.48 | *** join/#asterisk jeff_phillips (n=fircuser@m485336d0.tmodns.net) |
| 15:06.41 | *** join/#asterisk hesco (n=hesco@c-76-17-99-50.hsd1.ga.comcast.net) |
| 15:07.56 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 15:08.01 | hesco | as I rerun make menuselect, I seem to have Applications -> app_amd enabled. But using AMD() in a dialplan cerates an erroe on that extension. How would I troubleshoot this issue? ANy ideas? |
| 15:13.43 | *** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 15:13.44 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:16.22 | cesar_CR | hello guys I am in Central America, a costumer wants a PBX system that get calls from a toll-free number in the US, |
| 15:16.28 | cesar_CR | <PROTECTED> |
| 15:26.12 | tzafrir_laptop | hesco, core show application AMD |
| 15:26.17 | festr_ | 3C |
| 15:26.20 | festr_ | sorry |
| 15:35.45 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 15:39.11 | *** join/#asterisk matsk (n=matkar@c83-253-97-45.bredband.comhem.se) |
| 15:45.48 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 15:48.22 | *** join/#asterisk Badrobot- (i=Badrobot@cpe-76-173-233-75.socal.res.rr.com) |
| 15:53.53 | hesco | Your application(s) is (are) not registered |
| 15:53.53 | hesco | Command 'core show application AMD' failed. |
| 15:54.54 | hesco | do I put that in modules.conf? |
| 16:03.16 | tzafrir_laptop | do you use autoload? |
| 16:03.21 | tzafrir_laptop | hesco, ==^ |
| 16:03.40 | tzafrir_laptop | if not: yes, you should have an explicit lopad of app_amd.so |
| 16:04.03 | tzafrir_laptop | or maybe you have an explicit noload for it? |
| 16:07.38 | aniasis | Hello |
| 16:11.38 | tzafrir_laptop | jbot, tell aniasis about ask |
| 16:12.10 | aniasis | ?? |
| 16:13.22 | tzafrir_laptop | wonders if aniasis wants to say anything other than an occasional "Hello" or "Hi" :-) |
| 16:14.42 | *** join/#asterisk jstoker (n=jstoker@80-41-138-12.dynamic.dsl.as9105.com) |
| 16:15.57 | stevetotaro | bonjour |
| 16:17.13 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:17.46 | stevetotaro | hey you fakhir |
| 16:17.55 | fakhir | ello |
| 16:19.55 | aniasis | is there anyway to make free calls using asterisk? |
| 16:20.06 | stevetotaro | yes |
| 16:20.15 | stevetotaro | hack someone else's box |
| 16:20.33 | stevetotaro | you can always call between extenstions for free too |
| 16:20.36 | aniasis | so you still must pay for calls thru the internet? |
| 16:21.07 | stevetotaro | you could get an FXO and splice off someone's POTS line |
| 16:21.17 | aniasis | POTS? |
| 16:21.24 | stevetotaro | ~google |
| 16:21.25 | jbot | methinks google is at http://www.yahoo.com |
| 16:21.33 | stevetotaro | ~pots |
| 16:21.34 | jbot | [~pots] POTS (Plain Old Telephone Service) is the term for a common analog phone line service as is used world-wide. The "phone company" is called FXO (~fxo), and the user end-point (or phone) is called FXS (~FXS). POTS supports 1 channel, and possibly call-waiting, 3-way calling, CID, as signalled to the telco. |
| 16:22.27 | stevetotaro | cheapest calling method i am aware of for the states and canada is magicjack |
| 16:22.50 | stevetotaro | either by itselff or get the sip creds and use it in asterisk as a tunk |
| 16:23.17 | aniasis | Wait |
| 16:23.27 | stevetotaro | i hate to wait |
| 16:23.40 | aniasis | :) |
| 16:23.43 | stevetotaro | time is the only thing we can never get back |
| 16:23.58 | aniasis | I am just trying understand the concept |
| 16:24.10 | *** join/#asterisk mort___ (n=mort@user-54405220.wfd76b.dsl.pol.co.uk) |
| 16:24.11 | stevetotaro | ~book |
| 16:24.12 | jbot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:24.35 | aniasis | yes... |
| 16:24.36 | fakhir | the cost is to put calls onto the telephone network |
| 16:24.51 | aniasis | yes. |
| 16:25.00 | stevetotaro | that and or bandwidth |
| 16:25.20 | stevetotaro | that stuff isn't free even though you can mooch |
| 16:26.07 | aniasis | So I so to place a call from one PXS to PXS I must pay. |
| 16:26.32 | stevetotaro | what is a PXS? |
| 16:26.39 | aniasis | FXS |
| 16:26.50 | aniasis | FXS to FXS |
| 16:26.50 | fakhir | POTS! |
| 16:26.59 | *** join/#asterisk jmardonesk (n=jmardone@200-126-122-189.bk8-dsl.surnet.cl) |
| 16:27.07 | stevetotaro | FXS is just an extension usually |
| 16:27.44 | stevetotaro | if you have two FXS on the same PBX then you can call all you want |
| 16:27.54 | jmardonesk | hi all, the asterisk 1.6.0.6 is stable? where i can found a comparation about version 1.4 and 1.6? |
| 16:28.00 | stevetotaro | if you connect them over the internet, then you just pay for bandwidth |
| 16:28.11 | stevetotaro | 1.4 is not stable |
| 16:28.27 | stevetotaro | at least the latest releases |
| 16:30.13 | jmardonesk | and the 1.6 releases? |
| 16:30.20 | stevetotaro | hah |
| 16:30.23 | stevetotaro | beta |
| 16:30.31 | aniasis | okay so FXS to FXS is IP to IP? |
| 16:30.33 | stevetotaro | might be stable for you |
| 16:30.42 | stevetotaro | ~book |
| 16:30.43 | jbot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:30.54 | stevetotaro | aniasis, you need to read |
| 16:30.55 | fakhir | aniasis, not necessarily |
| 16:30.56 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 16:31.03 | fakhir | yeah you really need to read the book |
| 16:31.10 | jmardonesk | the last time that i install asterisk was i year ago, with the asterisk 1.4.21.2 |
| 16:31.46 | stevetotaro | i stil prefer 1.2 but..... |
| 16:32.24 | stevetotaro | my learning curve is going into freesswitch rather than 1.6 and dahdi |
| 16:34.34 | stevetotaro | General Growth Properties is looking like it is going under soon |
| 16:34.54 | stevetotaro | they own 200 malls is 44 states |
| 16:35.10 | aniasis | malls sucks |
| 16:35.11 | stevetotaro | bet BO will bail them out too |
| 16:35.26 | stevetotaro | i don't go to the mall either |
| 16:35.37 | stevetotaro | but one maill may employ thousands |
| 16:35.45 | aniasis | there are too many malls in america, everyone knows this |
| 16:35.51 | stevetotaro | HUGE profit centers and jobs |
| 16:36.08 | aniasis | if they were so profitable then they would not be going out of business |
| 16:36.24 | stevetotaro | called depression |
| 16:36.35 | stevetotaro | guess you haven't taken macro econ yet |
| 16:36.40 | aniasis | well then that means they aren't profitable. |
| 16:36.49 | aniasis | You can not be profitable and go under. |
| 16:36.56 | stevetotaro | nothing is, so just close it up but wallmart |
| 16:37.20 | aniasis | yes there are many profitable companies |
| 16:37.29 | stevetotaro | oh wait, all the unemployed ffrom the malls won't be albe to buy anything |
| 16:37.40 | stevetotaro | so wallmart will lose profit |
| 16:37.53 | stevetotaro | and all the other companies you say are profitable |
| 16:38.01 | stevetotaro | you need to read alot |
| 16:38.57 | stevetotaro | http://www.youtube.com/watch?v=Q2qDW34Fr64 |
| 16:39.03 | *** join/#asterisk bl4 (n=bl4qkuba@dsl5-ore-167.fiber.net) |
| 16:39.15 | aniasis | i won't argue about this. because if the company is not profitable then it is a bubble waiting to burst. so it is inevitable that it will fail. |
| 16:40.02 | stevetotaro | not much |
| 16:40.17 | stevetotaro | so US will fail |
| 16:40.20 | aniasis | if that mall property giant is going to go under, then it is likely because they attempted to grow too fast, simply being supported by investor dollars, instead of profit. |
| 16:40.23 | stevetotaro | glad i have my weapons |
| 16:40.41 | aniasis | the US will not fail because malls fail |
| 16:40.54 | stevetotaro | dominos buddy |
| 16:41.09 | aniasis | the economy just has to go into a different direction. |
| 16:41.48 | aniasis | there is no innovation in any country. |
| 16:42.15 | aniasis | so there is nothing to drive labor. consumerism drove labor in the US for the past two decades. |
| 16:42.46 | aniasis | and the US's consumerism is what moved the global economy. |
| 16:42.56 | stevetotaro | and.... |
| 16:43.18 | jaytee | the engine is sputtering and dying because it's running out of gas and there's no filling station in site |
| 16:43.21 | stevetotaro | does that stop the starving, robbing, riots? |
| 16:43.50 | aniasis | stevetotaro, no. but nothing will stop starving, robbing, and riots. |
| 16:44.15 | aniasis | stevetotaro, while you are arguing here with me you could be doing some profitable with your time. |
| 16:44.31 | stevetotaro | i am multi tasking |
| 16:44.35 | aniasis | sure. |
| 16:44.40 | stevetotaro | and it is not arguing, just waking you up |
| 16:44.54 | aniasis | you can't wake me up. |
| 16:44.59 | stevetotaro | guess not |
| 16:45.12 | aniasis | you should be trying to work with me instead of telling me something I already know |
| 16:45.22 | *** join/#asterisk eaxxae (n=x@unaffiliated/eaxxae) |
| 16:45.25 | stevetotaro | why would i work with you? |
| 16:45.29 | jaytee | "Wake up people! We've got less than inch of topsoil left!" - Tommy Lee Jones in Under Seige |
| 16:45.30 | aniasis | but you'd rather sit up and show how smart you are. |
| 16:45.34 | stevetotaro | you want the easy answer handed to you |
| 16:45.48 | stevetotaro | read, i was never on IRC asking for answers |
| 16:45.54 | stevetotaro | ~stevetotaro |
| 16:45.55 | jbot | methinks stevetotaro is an IRC nub |
| 16:46.10 | stevetotaro | i figure stuff out for myself |
| 16:46.15 | stevetotaro | i read |
| 16:46.21 | aniasis | stevetotaro, I'm not even interested in VOIP |
| 16:46.27 | jaytee | especially not asking how to properly spell Achievements on their website, http://www.totarotechnologies.com |
| 16:46.43 | stevetotaro | blow me jaytee |
| 16:46.44 | aniasis | And in fact I was wondering what is so good about Asterisk over Skype |
| 16:46.58 | stevetotaro | use skype |
| 16:47.17 | stevetotaro | you don't want voip but you want skype, lmao |
| 16:47.37 | aniasis | stevetotaro, see your joomla website sucks you should ask me to help you with this |
| 16:47.57 | stevetotaro | don't need to, i get plenty of work |
| 16:48.11 | aniasis | stevetotaro, I said I am not interested in learning VOIP. Only thing I wanted to know is why Asterisk is better than Skype. |
| 16:48.25 | stevetotaro | it isn't if you don't want to learn |
| 16:48.41 | stevetotaro | ~skype |
| 16:48.42 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, with most solutions being complex, complicated, and hack-ish . Digium's SkypeForAsterisk (see ~SkypeForAsterisk) is a new solution that is a cleaner non-dependent option. |
| 16:48.44 | aniasis | stevetotaro, then you can single-handlely save the WORLD ECONOMY, therefore you can save the WORLD. |
| 16:48.55 | aniasis | You would be like Superman! |
| 16:49.06 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 16:49.11 | stevetotaro | i would |
| 16:49.25 | stevetotaro | i would put people like ron paul in power |
| 16:49.34 | aniasis | Well then get to it buddy, because it seems like you are just sitting here arguing with me |
| 16:49.40 | aniasis | and being an asshole. |
| 16:49.55 | aniasis | ignores stevetotaro |
| 16:49.59 | stevetotaro | i cant help it if you "dont want to learn" |
| 16:50.56 | hesco | how can I reload only the modules.conf? |
| 16:51.17 | stevetotaro | type reload on the cli |
| 16:51.38 | aniasis | how can Asterisk be the future of telephony when it is not even peer-to-peer? |
| 16:51.44 | hesco | thanks |
| 16:51.53 | hesco | now to test it |
| 16:51.54 | stevetotaro | no problem |
| 16:51.56 | jaytee | aniasis, don't waste your time "tilting at windmills". He's just a nutcase pathological liar with delusions of grandeur, a bunker mentality of doom and gloom and can talk the talk but can't walk the walk. |
| 16:52.00 | hesco | that didn't seem to break it |
| 16:52.32 | hesco | it looked like that also reloaded the extensions, as well |
| 16:52.41 | stevetotaro | i ask for one example besides a lame website |
| 16:53.02 | stevetotaro | one lie? |
| 16:53.11 | aniasis | jaytee, who that stevetotaro guy? Oh you can tell he is a troll. |
| 16:53.46 | jaytee | he goes on and on about his black ops buddy and all this crap for hours. has several websites that are all lame. |
| 16:53.55 | *** join/#asterisk andrewn (n=andrew@70.36.140.13) |
| 16:54.12 | aniasis | jaytee, yeah that one I just saw was horrible! |
| 16:54.17 | stevetotaro | again, pointing to lame websites.... |
| 16:54.26 | stevetotaro | oh, there is proof |
| 16:54.35 | stevetotaro | you are the one stalking me obviously |
| 16:55.04 | stevetotaro | i have not or would not visit your website(s) because I really don't care about you.... LMAO |
| 16:55.17 | jaytee | no, I just clicked on a few links supplied by someone else in here that "has your number" :-) your reputation is well known around here |
| 16:55.38 | aniasis | jaytee, one of my colleagues setup an Asterisk box and virtualizes telephone numbers. I wanted to get a better understanding of how could I come in and do some quality on his business practices. |
| 16:55.40 | stevetotaro | because i have been doing asterisk for a very long tiime |
| 16:55.53 | [TK]D-Fender | [12:48]<aniasis>stevetotaro, I said I am not interested in learning VOIP. Only thing I wanted to know is why Asterisk is better than Skype. |
| 16:55.59 | stevetotaro | spoke at astricon about large call centers |
| 16:55.59 | [TK]D-Fender | aniasis: You are comparing apples and oranges |
| 16:56.09 | aniasis | [TK]D-Fender, yes I know. |
| 16:56.17 | [TK]D-Fender | aniasis: So why the pointless question? |
| 16:56.44 | aniasis | [TK]D-Fender, but I don't mean Skype by service, but by the P2P protocol |
| 16:56.59 | aniasis | Asterisk is a C/S based model correct? |
| 16:57.04 | [TK]D-Fender | aniasis: Asrterisk isn't a PROTOCOL either. So again, apples & oranges |
| 16:57.23 | aniasis | I never said it was a protocol. |
| 16:57.37 | aniasis | Does Asterisk use P2P? |
| 16:57.46 | [TK]D-Fender | aniasis: * is a telephony toolkit that can speak in several protocols. All of the VoIP ones are as a B2BUA |
| 16:58.07 | [TK]D-Fender | aniasis: use P2P? What exactly is that supposed to mean? |
| 16:58.44 | [TK]D-Fender | aniasis: On * can talk directly to another. Is that P2P? I can use a SIP phone with *. Since I can call from the * CLI with a headset, does that make * a CLIENT? |
| 16:58.52 | hesco | does reload know a graceful way of doing it without interrupting ongoing calls? |
| 16:59.06 | [TK]D-Fender | hesco: "reload" does not affect calls in progress |
| 16:59.13 | stevetotaro | reload should not drop calls |
| 16:59.26 | stevetotaro | but it has or had problems with queues |
| 16:59.35 | stevetotaro | you can reload specific modules |
| 16:59.42 | [TK]D-Fender | indeed |
| 16:59.46 | stevetotaro | type reload and hit tab a few times |
| 16:59.54 | [TK]D-Fender | aniasis: So, what are you getting at exactly? |
| 17:00.21 | aniasis | [TK]D-Fender, Look. I don't think you understand the question. If you are using an SIP phone with Asterisk and Asterisk is on the device which the SIP phone is accessing then that is the client. |
| 17:01.02 | stevetotaro | i think Fender understands very well |
| 17:01.24 | [TK]D-Fender | aniasis: Then * can be both a "client", and a "server" in so much that as far as SIP goes it is not a SWITCH, or ROUTER |
| 17:01.27 | aniasis | But if you connect directly to another SIP phone then without going through an arbitrary machine then it is Perr to Peer |
| 17:02.04 | [TK]D-Fender | aniasis: Well my * servers talk directly to SIP phones. So that makes it P2P by your definition, right? |
| 17:02.05 | aniasis | [TK]D-Fender, But if I want to make numbered calls then what? |
| 17:02.17 | [TK]D-Fender | aniasis: "numbered"? pardon? |
| 17:02.45 | aniasis | a call to a landline |
| 17:02.48 | *** join/#asterisk matsk (n=matkar@c83-253-97-45.bredband.comhem.se) |
| 17:02.57 | *** part/#asterisk jstoker (n=jstoker@80-41-138-12.dynamic.dsl.as9105.com) |
| 17:02.58 | [TK]D-Fender | aniasis: Yes, that would be "PSTN" you're looking for. |
| 17:03.27 | aniasis | and no it is not Peer-to-Peer if a call is being routed through you * server. |
| 17:03.56 | aniasis | So with PSTN you have to pay some communication network. |
| 17:04.11 | [TK]D-Fender | aniasis: Through? * IS the client in my sample. I go to * CLI and tell it to call out. BAM... a CLIENT |
| 17:04.40 | aniasis | you do this thru some sort of interface correct? |
| 17:04.56 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 17:05.01 | [TK]D-Fender | aniasis: And for PSTN, that does imply reaching to global telephony network and generally yes, someone is paying for that |
| 17:05.15 | aniasis | or are you calling a PSTN |
| 17:05.33 | [TK]D-Fender | aniasis: To get to the PSTN you can get sever kinds of links from telcos, ITSP's, etc |
| 17:05.51 | [TK]D-Fender | aniasis: Oh now you're being picky about WHO I'm calling for your definition of P2P? |
| 17:06.36 | [TK]D-Fender | aniasis: Please try to avoid corrupting the definitions of words... that tends to go very wrong in telecom circles |
| 17:06.38 | aniasis | [TK]D-Fender, you are trying to argue. I am trying to get an understanding |
| 17:06.56 | [TK]D-Fender | aniasis: You are, but you're clinging to a broken definition. |
| 17:07.15 | aniasis | because it is very important who you are calling |
| 17:07.29 | [TK]D-Fender | aniasis: So * can talk to a variety of devices. If you want it to use say a regular phone line in your room you'd need a hardware interface. |
| 17:07.51 | aniasis | of course. like those magic jack devices |
| 17:07.59 | [TK]D-Fender | aniasis: If you use an ITSP all you need is internet connectivity |
| 17:08.10 | [TK]D-Fender | aniasis: No. |
| 17:08.30 | [TK]D-Fender | aniasis: MagicJack is a FXS interface for you to use a PHONE with and goes to their ITSP |
| 17:08.40 | [TK]D-Fender | aniasis: you do not plug a PHONE LINE into it |
| 17:08.48 | [TK]D-Fender | aniasis: This is backwards. |
| 17:09.12 | [TK]D-Fender | aniasis: If you have a phone line from the telephone company in your room you need an FXO interface. |
| 17:09.18 | aniasis | But if I wanted to wire my home to an * server then what? |
| 17:09.36 | aniasis | an FXO interface that would do? |
| 17:10.04 | *** join/#asterisk jeff_phillips (n=fircuser@m485336d0.tmodns.net) |
| 17:10.10 | [TK]D-Fender | aniasis: When you say "your home", do you mean the line a telco give you, or instead the PHONES in your house? |
| 17:10.33 | aniasis | the jack in my home. |
| 17:10.45 | aniasis | yes the line that is installed be the telco |
| 17:11.10 | [TK]D-Fender | aniasis: Then yes you would need an FXO interface. |
| 17:11.26 | aniasis | and where would I place this interface? |
| 17:11.37 | hesco | thanks for the lead on the tab auto complete. |
| 17:11.40 | *** join/#asterisk romb_work (i=user@89.28.249.107) |
| 17:11.43 | hesco | that looks useful |
| 17:11.45 | aniasis | on that little gray box outside? |
| 17:12.45 | [TK]D-Fender | aniasis: location doesn't matter as long as it plugged in. Most people use basic PCI cards in their * boxes for this |
| 17:12.59 | stevetotaro | ciscoesque with the tab |
| 17:13.02 | romb_work | hello all |
| 17:13.04 | stevetotaro | it helps alot |
| 17:13.10 | stevetotaro | hi romb |
| 17:13.20 | [TK]D-Fender | aniasis: http://www.digium.com/en/products/analog/ <- some sample hardware |
| 17:13.28 | romb_work | any one know is Asterisk 1.6.0.6 support IMs without additional patches? |
| 17:13.32 | aniasis | [TK]D-Fender, yes. I was just visualizing at what point... |
| 17:13.40 | [TK]D-Fender | aniasis: Plenty of external gateways that would takee in your line and spit the call out over SIP, etc |
| 17:13.58 | [TK]D-Fender | aniasis: You'd plug it in just like you would a broing modem. |
| 17:14.03 | [TK]D-Fender | boring* |
| 17:14.05 | aniasis | [TK]D-Fender, but if I wanted to make a call from a cellphone to say aninternation number |
| 17:14.19 | jeff_phillips | just joined in the middle of this conversation, but anlasis: depending on what you are doing, i suggest unplugging all phones connected to that telco line in your house. |
| 17:14.37 | aniasis | excuse me international number |
| 17:14.48 | jeff_phillips | it gets confusing when you are using the line and the pbx does not know this so it barges in trying to use it as well. |
| 17:14.56 | jeff_phillips | hook all your phones in as extensions |
| 17:15.05 | aniasis | jeff_phillips, please stop. |
| 17:15.16 | jeff_phillips | ok sorry |
| 17:15.19 | *** join/#asterisk jstoker (n=jstoker@80-41-138-12.dynamic.dsl.as9105.com) |
| 17:15.21 | [TK]D-Fender | aniasis: How does your cellphone get to *? How would * get to "an international number"? |
| 17:15.45 | [TK]D-Fender | jeff_phillips: No one you want to jump into the middle of |
| 17:15.47 | aniasis | [TK]D-Fender, exactly. So I would have to 'call into' an * box |
| 17:15.59 | romb_work | **sorry |
| 17:16.08 | [TK]D-Fender | aniasis: well lets jsut say the microwaves aren't going to reach * by magic. |
| 17:16.11 | romb_work | any one know is Asterisk 1.6.0.6 support IMs *on SIP protocol* without additional patches? |
| 17:16.36 | aniasis | [TK]D-Fender, okay. Now we are at the same point. |
| 17:17.03 | aniasis | [TK]D-Fender, so now to make those calls internationally I would have to pay for some minutes. |
| 17:17.17 | [TK]D-Fender | aniasis: the question becomes "what do I really want to acheive and what ways can I do it". |
| 17:17.48 | [TK]D-Fender | aniasis: Generally you don't get to terminate to the PST for free. SOMEONE has to be paying, so what means are you looking to sue to terminate that call? |
| 17:17.48 | *** part/#asterisk jstoker (n=jstoker@80-41-138-12.dynamic.dsl.as9105.com) |
| 17:18.04 | aniasis | yes |
| 17:18.29 | aniasis | If you could point me to some resource then that would be helpful |
| 17:18.52 | *** join/#asterisk Badrobot- (n=Badrobot@cpe-76-173-233-75.socal.res.rr.com) |
| 17:18.59 | [TK]D-Fender | aniasis: I could point you to a dozen ITSP's, just as many telco's, etc..... |
| 17:19.18 | jeff_phillips | you can call your pots line from your cell, have * answer, then have it call out using a sip termination provider that has low rates to the destination country, but you would still be using mintues off your cell plan. unless the pots line is assigned as your myfaves or whatever on the cell plan |
| 17:19.20 | [TK]D-Fender | aniasis: Is that all you're looking for? Who'll terminate your call cheper? Rates will vary |
| 17:19.53 | aniasis | [TK]D-Fender, no because this is what my colleague already does. |
| 17:20.07 | [TK]D-Fender | aniasis: then what are you looking for? |
| 17:20.19 | aniasis | he has this service already up an running. I am just trying to get upto speed to what he is doing. |
| 17:20.32 | [TK]D-Fender | aniasis: Maybe you should simply ask HIM |
| 17:20.40 | stevetotaro | magicjack is awesome really |
| 17:20.57 | stevetotaro | by itself or with the sip cred hack |
| 17:21.09 | aniasis | [TK]D-Fender, Yes I will but. I am getting additional perspectives so I can perhaps bring some new ideas. |
| 17:21.48 | [TK]D-Fender | aniasis: Starting from near-zero in telecom and trying to bring "new ideas" toa guy who works professionally in the field? Scary. |
| 17:21.55 | jeff_phillips | it all depends on what countries you wish to call |
| 17:22.09 | aniasis | [TK]D-Fender, I am a genius of sorts. |
| 17:22.29 | [TK]D-Fender | aniasis: No doubt. |
| 17:22.50 | aniasis | [TK]D-Fender, well who cares. But thanks for your insight. |
| 17:23.07 | aniasis | even though it was nominal |
| 17:23.18 | [TK]D-Fender | aniasis: Hopefully you come toa clue about what it is you actually want to do. |
| 17:23.31 | romb_work | any one can help with SIP: MESSAGE or all ignoring me? |
| 17:23.46 | [TK]D-Fender | aniasis: So far we have a cell phone, an "international call", and a giant "gray are" in between |
| 17:23.53 | [TK]D-Fender | area* |
| 17:23.55 | aniasis | [TK]D-Fender, well really there is not much I can add to the process he is using. |
| 17:24.00 | *** join/#asterisk nix8n82 (n=nate@mo-65-41-196-62.sta.embarqhsd.net) |
| 17:24.07 | jeff_phillips | hence * as a wildcard. hehe |
| 17:25.35 | [TK]D-Fender | aniasis: Oh well. Let us know when you have a specific goal to actually achieve. |
| 17:26.31 | aniasis | [TK]D-Fender, Well lets rewind. |
| 17:26.53 | aniasis | to make an SIP to SIP call using Asterisk what would I need? |
| 17:27.16 | [TK]D-Fender | aniasis: Describe what is on each end |
| 17:27.36 | aniasis | two PCs connected through the internet |
| 17:27.54 | [TK]D-Fender | aniasis: 2 PC's and a machine running *. |
| 17:28.02 | aniasis | who is this tzafrir_laptop, that keeps PMing me? |
| 17:28.03 | [TK]D-Fender | aniasis: Which could be one of those 2 PC's |
| 17:28.12 | jeff_phillips | actually... two softphones and you dont even need * |
| 17:28.26 | aniasis | [TK]D-Fender, so one PC must have the * server running? |
| 17:29.01 | [TK]D-Fender | aniasis: No, I said COULD <- |
| 17:29.20 | aniasis | whatever |
| 17:29.34 | aniasis | anyway. 2 PCs connected through the internet. |
| 17:29.38 | [TK]D-Fender | aniasis: My wording has been very explicit. |
| 17:30.07 | [TK]D-Fender | aniasis: So to have 2 SIP devices call from one to the other via * requires those 2 devices, and a system running *. |
| 17:30.20 | aniasis | yes |
| 17:30.49 | aniasis | how would they connect to one another? |
| 17:31.12 | aniasis | they would have to have some sort of interface correct? |
| 17:31.20 | aniasis | to the * server. |
| 17:31.48 | [TK]D-Fender | aniasis: Via *, 1 places a call to * targeting an extension. * will choose to authenticate the call or not and if allowed it will enter the dialplan. Your dialplan would be configerd to match that extension and dial the other device. |
| 17:31.54 | nix8n82 | RTFM and a couple rfc docs it tells you everything |
| 17:32.26 | [TK]D-Fender | aniasis: no special hardware is required for this. Its just SIP in from one device, and SIP out the the other |
| 17:33.34 | aniasis | hmmm, well then specifically an instant messenger. how could you achieve this using * |
| 17:33.58 | [TK]D-Fender | aniasis: huh? |
| 17:34.42 | aniasis | similar to skype's services |
| 17:34.52 | [TK]D-Fender | aniasis: * does not support SIP IM, etc. no "texting" here. |
| 17:35.02 | nix8n82 | a lot of programming using agi and ami possiably |
| 17:35.44 | aniasis | hmmm |
| 17:35.44 | [TK]D-Fender | aniasis: * is a TELEPHONY toolkit, not an "MSN/AIM?IRC replacement" |
| 17:36.22 | aniasis | skype is a telephony/msn/aim/irc replacement |
| 17:36.46 | [TK]D-Fender | aniasis: Kudos to them. |
| 17:36.57 | stevetotaro | then use skype!!!!!!111111 (even though he is ignoring me......) |
| 17:37.44 | aniasis | [TK]D-Fender, but there has to be some application in using * for voice chat. |
| 17:37.46 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 17:38.08 | [TK]D-Fender | aniasis: "voice chat"? thats all * does. VOICE. |
| 17:38.25 | nix8n82 | anua how old are you? |
| 17:38.54 | nix8n82 | er aniasis |
| 17:40.01 | aniasis | 29 |
| 17:40.18 | mvanbaak | what does it matter how old someone is ? |
| 17:41.08 | *** join/#asterisk jeff_phillips (n=fircuser@m485336d0.tmodns.net) |
| 17:41.20 | aniasis | bbl |
| 17:41.24 | [TK]D-Fender | [13:35]<nix8n82>a lot of programming using agi and ami possiably <- No, * won't be doing any of the IM dirty work |
| 17:42.40 | nix8n82 | just curious if he was an ignorant kid or someone that needs a helment and a hand to hold. |
| 17:43.04 | [TK]D-Fender | nix8n82: Remeber the cat.... remember..... |
| 17:43.25 | stevetotaro | lol |
| 17:46.18 | nix8n82 | yeah I know * would not do any im work but just a cog in a speech to text app or some small paaaart in a javaaaaa oor wwweb app |
| 17:46.43 | hardwire | prefers to remember the milk |
| 17:48.07 | nix8n82 | any one else use a vnc and have trouple with multiple characters being sent for no appparant reason |
| 17:48.56 | jeff_phillips | He's got an imagination that will inspire him to achieve great things once he figures out what each tool does & how to use them. Give him time. |
| 17:50.48 | stevetotaro | i use VNC but no character problems |
| 17:51.01 | stevetotaro | mainly on windows though |
| 17:51.03 | [TK]D-Fender | jeff_phillips: Blind faith.... ask your doctor if religion is right for you! |
| 17:51.45 | stevetotaro | where never is heard, a discouraging world, and the skies are not cloudy all day.... |
| 17:51.55 | stevetotaro | word=word |
| 17:52.09 | stevetotaro | these netbooks mess up typing completely |
| 17:52.27 | stevetotaro | now i cannot type on a regular KB or the netbook |
| 17:53.04 | stevetotaro | anyone use eyeq? |
| 17:53.23 | *** join/#asterisk jstoker (n=jstoker@88-107-192-59.dynamic.dsl.as9105.com) |
| 17:53.24 | nix8n82 | yeah I'm kinda getting use to mine. I got one of those flexiable rollup keyboards and I can almost use a full size keyboard again |
| 17:53.25 | stevetotaro | amazing program, i doubled my reading speed at same retention |
| 17:54.17 | nix8n82 | no kidding how much does it cost? |
| 17:54.29 | stevetotaro | well i am not sure |
| 17:54.41 | jeff_phillips | clearly his own faith is not blind. he sees his friend setup something cool/useful, sees potential in the tools being able to achieve even more, and wants to learn about them. thats not blind faith. thats pursuing a goal. |
| 17:54.45 | stevetotaro | i rarely "buy" software..... |
| 17:55.16 | stevetotaro | jeff, you missed the part where he said he "didn't want to learn about voip" |
| 17:55.28 | [TK]D-Fender | jeff_phillips: Not quite.... he actually does not understand any of the individual bits involved or what role * plays in them. He also has no stated goal. |
| 17:55.29 | jeff_phillips | oh. |
| 17:55.57 | stevetotaro | [12:48]<aniasis>stevetotaro, I said I am not interested in learning VOIP. Only thing I wanted to know is why Asterisk is better than Skype. |
| 17:56.25 | nix8n82 | even I read a few docs about the program before I came in here and started asking questions |
| 17:56.59 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 17:57.17 | stevetotaro | i didn't come here until late 08 |
| 17:57.27 | jeff_phillips | well, i dont really care to learn about how voip works in and of itself either, but learning how to work with it and put it to practical use is another thing, and that im interested in |
| 17:57.49 | stevetotaro | you are a good man jeff |
| 17:58.03 | stevetotaro | better than me |
| 17:58.34 | stevetotaro | unfortunately you kinda need to know how it works to put it to practical use |
| 17:58.38 | stevetotaro | or just use skype |
| 17:58.55 | stevetotaro | or my favorite, magicjack!!!! |
| 17:59.00 | jeff_phillips | we are all newbies at something, and the experts were once newbies at their current area of expertise too |
| 17:59.11 | jaytee | haha, magicjack |
| 17:59.13 | stevetotaro | but some read a bit |
| 17:59.42 | stevetotaro | some search for the answers instead of asking them to be silver spoon fed |
| 17:59.46 | jeff_phillips | once he realizes what he wants to do, he will learn about whatever he has to learn to make it happen |
| 18:00.04 | jeff_phillips | oh i agree there |
| 18:00.07 | jaytee | SIP Demystified is an excellent book for anyone who isn't of the "I'm not interested in learning about VOIP" mindset |
| 18:00.09 | [TK]D-Fender | stevetotaro: I swear I just want to smack people every time I hear : MagicJack, Skype, Vonage, Google Voice, GrandCentral, etc. |
| 18:00.38 | jaytee | MagicJack, Skype, Vonage, Google Voice, GrandCentral, etc. :-) |
| 18:00.43 | [TK]D-Fender | jaytee: Everything your typical newb needs to know can fit on a page. |
| 18:00.48 | *** join/#asterisk mahlon (i=mahlon@martini.nu) |
| 18:00.55 | nix8n82 | anyone know of an opensource speech to text tool that is somewhat easy to use > |
| 18:01.09 | stevetotaro | magicjack aint so bad though if you hack out the sip creds and use them with * |
| 18:01.25 | nix8n82 | ? |
| 18:01.28 | jaytee | yeah, there's a few people have managed to use it successfully |
| 18:01.30 | stevetotaro | $20/yr for unlimited |
| 18:01.38 | stevetotaro | i have four of them |
| 18:01.56 | stevetotaro | all sip "trunks" on an ast box |
| 18:01.59 | [TK]D-Fender | stevetotaro: thats the problem... every stupid faggot kiddie how wants to be "cool" and try to break their terms of service, etc because they can't afford the very inexpensive rates already available. |
| 18:02.16 | jeff_phillips | yeah but if a lot of people did that the company would go out of business |
| 18:02.25 | [TK]D-Fender | stevetotaro: Oh yes... and don't forget to leave those using the words "sip trunks" off my "hate list" :) |
| 18:02.29 | jaytee | an open source speech to text tool that's easy to use? hmmm, lemme think a moment......... Nope, I got nuthin! |
| 18:02.32 | stevetotaro | i don't really think it is viable anyways |
| 18:02.46 | *** join/#asterisk matsk (n=matkar@c83-253-97-45.bredband.comhem.se) |
| 18:03.03 | *** join/#asterisk jstoker (n=jstoker@88-107-192-59.dynamic.dsl.as9105.com) |
| 18:03.10 | carrar | jaytee, what a festival of a time! |
| 18:03.16 | stevetotaro | i am friends with the former CTO of Sunrocket |
| 18:03.27 | stevetotaro | met with him the weekend after they went down |
| 18:03.39 | jaytee | carrar, um isn't that going in the opposite direction? text to speech? |
| 18:03.45 | jeff_phillips | it definately is a flawed business plan. they base it off of average usage statisitcs of what was available before their product |
| 18:03.57 | nix8n82 | how about one that is kind of a challange and proven to work with asterisk generated sound files? |
| 18:03.58 | carrar | yeah |
| 18:04.05 | carrar | I'm just injecting random comments |
| 18:04.31 | carrar | fillin up dah tubes |
| 18:04.40 | [TK]D-Fender | carrar: not under my radar you aren't! :p |
| 18:04.48 | stevetotaro | pr0n fills the tubes |
| 18:05.04 | carrar | Always someone out to spoil my fun!! |
| 18:05.06 | [TK]D-Fender | stevetotaro: Sometimes all of them at once... but thats illegal in several states... |
| 18:05.07 | stevetotaro | surprised voip can get through being a RTP and all |
| 18:06.02 | LotR | didn't the guys who did festival do the reverse as well? the name is eluding me right now though :( |
| 18:06.09 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 18:06.22 | stevetotaro | festival is SO BAD |
| 18:06.28 | Dovid | hi. has anyone ever tried to work with an asterisk server behing an ISA server ? |
| 18:06.37 | stevetotaro | my commodore 64 sounds about the same |
| 18:06.44 | jaytee | Cepstral is decent |
| 18:06.49 | jaytee | although it's not free |
| 18:06.57 | stevetotaro | cepstral is really good |
| 18:07.12 | stevetotaro | it's free if you use sox to chop the audio |
| 18:07.17 | jaytee | I use it for taking spanish text and converting it to spanish voice prompts for our IVR |
| 18:07.18 | stevetotaro | or don't mind the nagging |
| 18:07.19 | [TK]D-Fender | nix8n82: Look at LumenVox |
| 18:07.46 | jaytee | lumenvox isn't easy to use for most people |
| 18:08.22 | [TK]D-Fender | jaytee: "most people" shouldn't be involved in telephony more than buying a cellphone |
| 18:08.32 | jaytee | and it's not really speech to text although you could code it but the dictionary is limited even with the full version to 12000 words. there's over 540,000 words in the english language. |
| 18:08.43 | nix8n82 | i know of sphinx but i also read where it isn't good because itt wasn't design for the 8000khz |
| 18:09.03 | [TK]D-Fender | sphinx(ter) |
| 18:09.17 | stevetotaro | 8khz is a standard POTS line, no? |
| 18:09.24 | [TK]D-Fender | stevetotaro: yup |
| 18:09.45 | nix8n82 | that what I thought |
| 18:09.52 | jaytee | I think he's looking for something like Dragon Dictate or the IBM branded version of the same product but that requires voice recognition training of the app, it's not free and I don't believe there's a linux version of it. |
| 18:10.01 | stevetotaro | lesson learned today |
| 18:10.15 | stevetotaro | avoid dollar store european coffee |
| 18:10.35 | stevetotaro | the water does not drain through it quickly and you get a big mess to clean up |
| 18:10.38 | jaytee | I buy Jamaican Blue Mountain green beans and roast my own |
| 18:11.18 | nix8n82 | you should use a press if you want good coffee anyway |
| 18:11.35 | stevetotaro | i am not a coffee snob |
| 18:12.01 | jaytee | and my Cuisinart burr grinder can grind 18 different levels of grind from powder to "chunk" |
| 18:12.03 | stevetotaro | i REALLY like nescafe in the little packets |
| 18:12.11 | nix8n82 | k but your onlyy cheating yourself |
| 18:12.12 | stevetotaro | can't seem to find them here in the states though |
| 18:13.21 | stevetotaro | why do birds keep flying into windows over and over |
| 18:13.42 | stevetotaro | this is the freaking forth day this robbin is poinding my kitchen window |
| 18:13.54 | stevetotaro | i am just going to open it and let him in |
| 18:14.13 | jeff_phillips | maybe the bird wants your coffee |
| 18:14.27 | stevetotaro | i am sure my pit bull would want the bird |
| 18:14.46 | stevetotaro | talk about a bull in a china shop |
| 18:14.59 | jeff_phillips | see, its a win win senario. you dont like the coffee, but do like the dog |
| 18:15.13 | stevetotaro | i love the dog |
| 18:15.20 | stevetotaro | that's my homeboy |
| 18:16.03 | stevetotaro | he is semi retired at 13 years though |
| 18:16.12 | jeff_phillips | well there ya go. let the bird have the coffee you dont like, and your dog will be happy to get the bird |
| 18:16.19 | stevetotaro | still a puppy at heart but has to recharge alot more |
| 18:17.13 | [TK]D-Fender | [14:14]<stevetotaro>talk about a bull in a china shop <-- Mythebusters completely disproved this |
| 18:17.36 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 18:18.13 | stevetotaro | missed that one |
| 18:18.30 | stevetotaro | but kick the bull in the balls and i am sure he might break something |
| 18:18.50 | tzafrir_laptop | hmm... the phrase used in Hebrew is "an elephant in a china shop". Good thing them Myth Busters didn't get to disprove it yet |
| 18:19.16 | jeff_phillips | you guys will think im crazy, but i have an idea for using uucp with speech recognition & text to speech software on an asterisk based voice mail system without ip access |
| 18:19.52 | tzafrir_laptop | jeff_phillips, uucp as in Unix to Unix CoPy? |
| 18:20.02 | jeff_phillips | yeah |
| 18:20.23 | tzafrir_laptop | why would anybody use it nowadays? |
| 18:20.52 | jeff_phillips | it would be for part of a disaster response tool kit |
| 18:21.07 | tzafrir_laptop | zeroconf |
| 18:24.01 | jeff_phillips | i want to make a single kit that can be thrown in a vehicle and quickly setup as a mobile commications command center integrating all availalble forms of communication - phone networks (landline and wireless), ip, voip, amature radio both voice and packet radio types of things, and satellite gear all in one box... |
| 18:24.43 | jeff_phillips | with a pc that runs a fancy database with mapping stuff and most importantly, a system people can check in to and leave messages for one another |
| 18:25.52 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 18:26.05 | jeff_phillips | so say a major city is evacuated, you show up at a safe location used as a makeshift shelter, the phones are out, but you put your regular home/cell number in this system and a message as to your status/location |
| 18:26.51 | jeff_phillips | the box, using whatever means of communications available, will store and forward this information to its peers (similar kits elsewhere) |
| 18:27.50 | stevetotaro | my mentor has vsat in his truck.... |
| 18:28.09 | jeff_phillips | eventually a peer with working connectivity will be reachable by your friends and family who can call into a hotline number and find out that you are ok or leave messages for you to receive |
| 18:28.25 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 18:28.55 | stevetotaro | OpenBTS, VSAT, 2 meter HAM Radio, app-rpt |
| 18:29.26 | jeff_phillips | the idea is that the entire system would be automated as much as possible for self discovery of its peer systems and the end users would only have to know how to use a regular touch tone phone and know each others normal phone number |
| 18:29.51 | jeff_phillips | stevetoaro: sounds like hes got a pretty sweet rig |
| 18:30.42 | stevetotaro | throw some hot air balloons with wifi up in the air |
| 18:30.48 | stevetotaro | or helium |
| 18:30.56 | stevetotaro | in a meshed network |
| 18:31.11 | jeff_phillips | really what i want to do is integrate all these technologies with a database that will propagate itself to answer a lot of the peoples questions before they are asked |
| 18:31.27 | stevetotaro | psychic? |
| 18:31.35 | stevetotaro | i don't think we are there yet..... |
| 18:31.43 | jeff_phillips | where is my family, are they ok? where can i go? what can i do to help? |
| 18:31.54 | stevetotaro | GPS |
| 18:32.15 | jeff_phillips | when the ptsn and internet break down those questions are in one location and the answers are in another. |
| 18:32.29 | stevetotaro | HAM radio |
| 18:32.55 | jeff_phillips | so much effort is expended trying to connect the two that could be better used in helping people effected by the disaster |
| 18:33.04 | stevetotaro | HAM was used to a great degree on 9/11 |
| 18:33.22 | stevetotaro | that is what first responders are for |
| 18:33.24 | stevetotaro | and FEMA |
| 18:33.26 | jeff_phillips | yes, all of those technologies play an important roll in disaster situations |
| 18:33.30 | stevetotaro | they use two way radio |
| 18:33.59 | stevetotaro | best thing you can do is bring food, water, gas |
| 18:34.58 | *** join/#asterisk matsk (n=matkar@c83-253-97-45.bredband.comhem.se) |
| 18:35.21 | jeff_phillips | i see the flexibility of asterisk as a way of integrating essentially every tool you can think of with the most common denominator in terms of a user interface everyone is famier with: verbal speech and a person's phone number |
| 18:35.50 | *** join/#asterisk matsk (n=matkar@c83-253-97-45.bredband.comhem.se) |
| 18:36.01 | stevetotaro | jeff: hint, that is why it is called "Asterisk" |
| 18:36.10 | stevetotaro | or "Wildcard" |
| 18:36.17 | jeff_phillips | i know |
| 18:36.37 | jeff_phillips | i just, want to put my tinker toy project together |
| 18:36.47 | stevetotaro | mine is already together |
| 18:36.56 | stevetotaro | maybe we can collaberate a bit |
| 18:37.14 | jeff_phillips | perhaps. |
| 18:37.48 | stevetotaro | http://www.first-notification.com/ |
| 18:37.59 | stevetotaro | yes jaytee, it is the same jooma template |
| 18:38.19 | stevetotaro | i already have the backend tech but it is not online with this front end yet |
| 18:39.05 | jaytee | I've just started messin with perl and php. man, "learnin's hard!" |
| 18:39.49 | mvanbaak | and php is bad |
| 18:39.58 | *** join/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 18:40.03 | jaytee | mvanbaak, what do you use? |
| 18:40.10 | LotR | mvanbaak: lots of people say that about perl too :) |
| 18:40.11 | stevetotaro | jaytee how about this custom joomla site http://dev.first-notification.com/website/ |
| 18:40.14 | mvanbaak | php, python and C |
| 18:40.30 | mvanbaak | jaytee: ^^ |
| 18:41.01 | jaytee | stevetotaro, that's damn slick! |
| 18:41.08 | jeff_phillips | stevetotaro: looks like a pretty useful service |
| 18:41.19 | mvanbaak | some of our core apps are written in php, and rewriting them in some other language takes too much time, but php is bad |
| 18:41.28 | jaytee | I like the flyover changes, that's a nice feature |
| 18:41.59 | stevetotaro | it is still joomla which is nice |
| 18:42.02 | hardwire | blah |
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| 18:42.56 | mvanbaak | the flyover should get some delay. this is anoying |
| 18:43.17 | stevetotaro | well it's a prototype of a stalled project |
| 18:43.26 | stevetotaro | but the whole backend is in place |
| 18:43.31 | jaytee | brief is spelled incorrectly |
| 18:43.45 | stevetotaro | i didn't do any of the "nice" site |
| 18:43.46 | jeff_phillips | cool |
| 18:43.53 | jaytee | i before e except after c, with some exceptions |
| 18:44.48 | mvanbaak | stevetotaro: creating a flyover thingie like that takes roughly 20 lines of code, and that's it |
| 18:44.52 | mvanbaak | nothing special |
| 18:45.22 | stevetotaro | chan_mobile is something special |
| 18:45.28 | mvanbaak | ok, maybe 50 if you add all browsers to it |
| 18:45.29 | stevetotaro | if it coud only do multiple phones |
| 18:45.59 | stevetotaro | i am no dev whatsoever |
| 18:46.13 | stevetotaro | i can read a bit of code, that is it |
| 18:46.19 | mvanbaak | create some divs, put content in them, hide them all by default, and add a js function that hides one div and shows another when you hoover a specific element |
| 18:46.27 | mvanbaak | basic js stuff |
| 18:46.47 | mvanbaak | and if you add a framework like jquery it's even easier |
| 18:47.04 | stevetotaro | first priority will be to get everything online |
| 18:47.31 | stevetotaro | then spelling and looks |
| 18:47.38 | stevetotaro | then profit!!!! |
| 18:47.58 | stevetotaro | oh, |
| 18:48.08 | stevetotaro | i forgot the paying customers part..... oooops |
| 18:48.56 | mvanbaak | ew, this page you pasted is even done in tables .... |
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| 18:49.31 | mvanbaak | and the effect is from dreamweaver |
| 18:49.38 | stevetotaro | which one the dev or the original? |
| 18:49.39 | mvanbaak | MM_preloadImages() |
| 18:49.52 | mvanbaak | dev.first-notification.com/website/ |
| 18:49.54 | mvanbaak | that one |
| 18:50.07 | stevetotaro | yeah, like i said, so long as it works and looks ok i could give a damn |
| 18:50.28 | mvanbaak | have you tested in safari 4 and ie8 ? |
| 18:51.17 | stevetotaro | no, it is totally beta |
| 18:51.27 | stevetotaro | i am concerned with the backend |
| 18:51.41 | stevetotaro | i have and will pay for the front end |
| 18:52.02 | mvanbaak | yup. backend is most important :) |
| 18:52.27 | mvanbaak | the backend is done in joombla as well ? |
| 18:53.35 | stevetotaro | no, it is C |
| 18:53.49 | stevetotaro | a few asterisk modules actually |
| 18:54.08 | mvanbaak | looks like a cool project to work on |
| 18:54.09 | stevetotaro | but i guess i cannot say asterisk since other code has changed |
| 18:54.54 | stevetotaro | i have stalled on it |
| 18:55.10 | stevetotaro | need some steam or someone to invigorate me |
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| 19:04.05 | jaytee | this sucks, it's snowing out! |
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| 19:24.11 | surfdue | jw, is there a php gui for condiguring the pbx |
| 19:25.58 | [TK]D-Fender | surfdue: What does it matter what language a GUI is written in? |
| 19:27.44 | surfdue | [TK]D-Fender: dosnt i can just code in php, do you have a gui that can allow configuration of the entire asterisk |
| 19:28.33 | [TK]D-Fender | surfdue: There are many GUI's out there, go take a look at #freepbx |
| 19:29.09 | [TK]D-Fender | surfdue: And keep in mind that "configuring the entire asterisk", means doing things ITS way and if you don't like it expect it to put up a fight. |
| 19:30.07 | surfdue | :} |
| 19:30.12 | surfdue | [TK]D-Fender: thanks for your help! |
| 19:39.29 | stevetotaro | anyone really try out druid? |
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| 19:43.32 | tzafrir_laptop | did |
| 19:43.38 | tzafrir_laptop | quite nice |
| 19:44.07 | tzafrir_laptop | data structures generally saner than those of FreePBX, and user interface is actually nicer |
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| 19:51.57 | phr3ak | what is freepbx? |
| 19:52.31 | phr3ak | is it asterisk clone or totally different? |
| 19:53.04 | tzafrir_laptop | It is something that runs on top of Asterisk |
| 19:53.29 | tzafrir_laptop | Asterisk is more of a PBX toolkit than a complete PBX |
| 19:53.50 | tzafrir_laptop | it's highly programmable and very flexible |
| 19:54.32 | tzafrir_laptop | FreePBX, asterisk-gui, Druid etc. try to create a complete PBX on top of Asterisk |
| 19:56.07 | phr3ak | anyone could help me? i'd like to create an extension to dialout |
| 19:56.35 | phr3ak | now i'm using this: exten => 601,1,Dial(Modem/modem:5551212) |
| 19:57.04 | phr3ak | but it's only one fix phone number |
| 19:57.34 | drmessano | O.o |
| 19:58.56 | [TK]D-Fender | drmessano: http://www.voip-info.org/wiki/view/Asterisk+Modem+channels |
| 19:59.18 | [TK]D-Fender | phr3ak: then maybe you should be using a VARIABLE number |
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| 20:01.49 | phr3ak | thank you |
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| 20:20.29 | surfdue | will installing asterisk and freepbx will it mess with cpanel :/ |
| 20:23.13 | [TK]D-Fender | surfdue: it has nothing to do with cpanel |
| 20:24.33 | surfdue | [TK]D-Fender: i cant do chown -R asterisk:asterisk /var/lib/php/session/, it will cause my cpanel to die |
| 20:24.53 | surfdue | that dosnt make any sense, why do i have to make my entire system run under the asterisk username this should its own little folder. |
| 20:24.53 | whitehat | can anyone point to to info on Bell Canada's tech info regarding inbound lines and voip issues. thank you. |
| 20:25.42 | [TK]D-Fender | whitehat What "info" about what "issues"? Sorry... could you be a little more vague please..... |
| 20:26.26 | surfdue | [TK]D-Fender: can i make asterisk NOT require me to edit my httpd shit, |
| 20:26.45 | [TK]D-Fender | surfdue: Asterisk doesn't |
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| 20:27.06 | surfdue | [TK]D-Fender: freepbx i guess then ..nvm |
| 20:28.21 | [TK]D-Fender | surfdue: Correct. There may be other ways of installing it that are less invasive, but that is THEIR problem, not ours |
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| 21:07.33 | jaytee | waves bye |
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| 21:47.20 | skyfer | Anyone have any ideas why echo "select 1" | isql -v asterisk-connector works, but "odbc show" in CLI shows no DSN entries? |
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| 21:55.52 | [TK]D-Fender | skyfer: Because general ODBC is setup fine and the Asterisk side isn't |
| 21:57.00 | skyfer | hmm, ok - so how can I perform troubleshooting on the Asterisk side? |
| 21:57.19 | skyfer | The res_odbc.so module has been loaded |
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| 21:59.30 | skyfer | Its "Use Count" is 0, though |
| 21:59.46 | skyfer | I don't know if that's important in this context or not |
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| 22:02.45 | aniasis | i'm back |
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| 22:51.01 | kimo_sabe | so, zttest, a reliable indication of timing problems or not? |
| 22:51.57 | kimo_sabe | I've got a PRI that's resetting and dropping all call a couple times a day and I'm trying to figure out why |
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| 23:31.18 | [TK]D-Fender | kimo_sabe: Look for IRQ misses, etc. "cat /proc/interrupts" |
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| 23:51.27 | desdesdesdes | hi there i have problem with dtmf on alalog cordless phones, anyone know how to increase the waiting for dtmf when doing a transfer using flash? |
| 23:51.53 | CrashSys | set relaxdtmf=yes |
| 23:53.08 | desdesdesdes | this would apply to zapata.conf? |
| 23:53.14 | CrashSys | yes |
| 23:53.18 | CrashSys | or sip.conf |
| 23:54.26 | desdesdesdes | only have problem on fxs extensions linked with xorcom fxs gateway, the problem is with the cordlessphones delay when sending digits |
| 23:56.47 | CrashSys | all I have is try setting relaxdtmf=yes or no |
| 23:56.50 | CrashSys | see if that helps |
| 23:56.54 | CrashSys | never messed with a xorcom |
| 23:57.21 | jaytee | on some gateways and ATAs there's a parameter for short and long interdigit timeout. might try adjusting that |
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| 23:57.52 | jaytee | and some sip phones with their own dialplans as well have that timeout feature |
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