| 00:01.54 | [TK]D-Fender | BAI BAI |
| 00:10.57 | roxdragon | BAY [TK]D-Fender |
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| 00:44.55 | s34n | I've been away from asterisk for a while, and am dipping back in. |
| 00:45.08 | v1s | is there any cmd, var or otherway to report how long a call was connected? |
| 00:45.17 | s34n | I'm looking for something like astyerisk realtime (from the 1.2 days) |
| 00:45.58 | s34n | There's probably some really common way to do this now |
| 00:46.22 | carrar | v1s, look in your CDR |
| 00:46.31 | s34n | But I don't want to have to parse a conf file, edit it, save it, and restart for every little change |
| 00:46.35 | bougyman | xml_cdr for more detail |
| 00:46.38 | s34n | I need something more dynamic |
| 00:46.39 | bougyman | oh woops, nm. |
| 00:46.56 | carrar | s34n, put your CDR's into a db like postgres |
| 00:47.02 | s34n | I know there is ldap |
| 00:47.10 | carrar | be smrt |
| 00:47.17 | s34n | carrar: I'm not worried about CDR. |
| 00:47.25 | carrar | You asked |
| 00:47.31 | s34n | v1s probably is, though |
| 00:47.38 | carrar | oh |
| 00:47.40 | carrar | hahah |
| 00:47.42 | carrar | yes |
| 00:47.49 | carrar | like I said, v1s ;) |
| 00:47.53 | s34n | :) |
| 00:47.55 | carrar | heh |
| 00:48.28 | s34n | carrar: but I would love to put extensions, voicemail, sip.conf, etc. in pg |
| 00:48.38 | carrar | s34n, you can put a lot of the config in a db also |
| 00:48.41 | carrar | like postgres! |
| 00:48.58 | carrar | you can put all that in a db |
| 00:49.00 | s34n | where do I read up about that? |
| 00:49.06 | carrar | The INTERNET |
| 00:49.10 | s34n | :) |
| 00:49.13 | v1s | thanks |
| 00:49.22 | carrar | download latest |
| 00:49.33 | carrar | sift through the provided docs |
| 00:49.39 | carrar | do a little googling |
| 00:49.47 | carrar | and you are good to go |
| 00:51.01 | s34n | yeah. great. |
| 00:51.18 | s34n | it looks like extconfig is a good place to start maybe |
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| 00:57.51 | s34n | what is a "family" in asterisk lingo? |
| 00:58.20 | [TK]D-Fender | s34n: Ask DAHDI |
| 00:58.53 | s34n | [TK]D-Fender: how? |
| 00:59.17 | [TK]D-Fender | s34n: Perhaps you should be more specific |
| 01:01.08 | s34n | http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions talks about "family" in extconfig.conf and in extensions.conf |
| 01:01.29 | s34n | for instance, switch => Realtime/[context]@[family][/options] |
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| 01:31.59 | jamko | you really only need switch=> realtime |
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| 02:15.58 | s34n | jamko: I'm wondering what "family" refers to there |
| 02:19.46 | jamko | the asterisk functions stored in the db are the "families" .. the families are referenced in extconfig .. For example in extconf.conf, if you had, extensions.conf=>mysql,asterisk,static_realtime_extensions |
| 02:20.45 | jamko | that is telling asterisk, to read the static extensions file, from mysql, in a database called asterisk, in the table called static_realtime_extensions |
| 02:21.02 | jamko | or |
| 02:22.03 | jamko | in extconfig.conf you could have, extensions=>mysql,asterisk,realtime_extensions |
| 02:23.02 | jamko | which is telling asterisk to read the extensions in realtime through from the mysql database "asterisk" from table "realtime_extensions" |
| 02:26.34 | jamko | you can have some families be "dynamic realtime" and others "static realtime" .... static realtime is specified by using the name of the config file, instead of the family. |
| 02:27.53 | jamko | technically the term "family" might only apply to dynamic realtime, since you are actually specifying the .conf file name when using static. Not completely sure on that part of it.. |
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| 03:29.43 | cabrel | when using a PIMG unit that has 8 ports connected to 8 station ports on a standard TDM PBX, does that translate to 8 concurrent calls can be made (assuming the trunk can support that) from the voip side? |
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| 04:35.08 | WIMPy | In which case(s) does DIALSTATUS get set to CANCEL? |
| 04:36.01 | [TK]D-Fender | WIMPy: "core show application dial" |
| 04:38.34 | WIMPy | Err. I only see it as possible result there. No forther information. |
| 04:39.01 | [TK]D-Fender | WIMYou should see an option that has ab obvious potential outcome like that... |
| 04:40.14 | WIMPy | d? |
| 04:45.40 | WIMPy | Not sure what DIALSTATUS would be good for, when jumping to another extension, but there isn't anything that strikes me as reallt obvious here. |
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| 09:15.15 | shamelessn00b | hmm |
| 09:15.18 | shamelessn00b | quite day today |
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| 12:02.31 | v1s | does this GotoIf($[${RAND(0,100)} < 95]?s,6) mean to do s,6 only 5% of the time? or is it mean do it 95%? thanks |
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| 12:26.44 | wdoekes2 | v1s: 95% |
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| 13:40.32 | v1s | thanks |
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| 13:58.41 | v1s | I have a provider that limits its calls to 5mins and I would like to setup some kinda of message or beep as a warning before the times up on the current call. Can some one tell me how or point me in the right direction. |
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| 14:00.14 | [TK]D-Fender | v1s: "core show application dial" <--- |
| 14:02.01 | v1s | [TK]D-Fender: thnks |
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| 14:04.15 | Gary_B | where does the avaya ip 403 hw box plug into the network so it can be configured from software |
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| 14:05.17 | [TK]D-Fender | Gary_B: Try asking in #avaya .... |
| 14:06.28 | [TK]D-Fender | Gary_B: Though by the way you worded that it sounds like the anser is "by its ethernet jack". |
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| 14:19.15 | v1s | is IAX the best way to connect to ast servers together? Or is SIP just as good? or something better? |
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| 14:20.52 | [TK]D-Fender | v1s: Depends. Use IAx if you NEED it. Otherwise I recommend SIP |
| 14:22.18 | v1s | [TK]D-Fender: what would warrant NEEDing it? There is clean path between the servers. One of them is on a slower connection is all. |
| 14:23.50 | [TK]D-Fender | v1s: Bandwith is one reason if you are passing multiple calls at the same time between them. Or if you are in a NAT-Hostile environment |
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| 17:20.25 | drenda81 | Hi guys |
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| 17:21.02 | drenda81 | I've a small problem in setting caller id with Asterisk 1.6.2 |
| 17:21.11 | drenda81 | I hope someone can help me |
| 17:21.38 | WIMPy | Not unless you tell us what it is. |
| 17:21.47 | drenda81 | if I set the caller id as this: exten => s,n,Set(CALLERID(all)="Name"<0426123456>) |
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| 17:22.16 | drenda81 | on my SIp phone Lynksys SPA942 I see the name "Name" but as number I read the number of my phone |
| 17:22.25 | drenda81 | and not the number that I've set (0426123456) |
| 17:23.05 | qwerty | hi there, anyone have binary codec_g729 for freebsd 7 x64 ? |
| 17:23.12 | WIMPy | I guess there should be a space, but I'm not sure it works like that at all. |
| 17:23.40 | WIMPy | If you set name and num seperately you are on the safe side. |
| 17:25.16 | drenda81 | WIMPy: So you tell me to use 2 lines as this: exten => s,n,Set(CALLERID(name)="Name") and exten => s,n,Set(CALLERID(num)=0426123456) ? |
| 17:26.24 | WIMPy | That will definitely work, but you shouldn't quote strings. |
| 17:27.48 | drenda81 | ok, I'm trying with your suggestions... |
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| 17:29.07 | csnook | Is there something magic about #asterisk that prevents X-Chat from reconnecting to it when I resume my laptop? |
| 17:34.35 | [TK]D-Fender | csnook: the fact FreeNode thinks you're reg'd elsewhere |
| 17:34.44 | drenda81 | WIMPy: it's seems not work: [Sep 25 19:34:09] -- Executing [s@tasto2:1] Set("SIP/6554448376-00000018", "CALLERID(name)=Name of test") in new stack |
| 17:34.44 | drenda81 | [Sep 25 19:34:09] -- Executing [s@tasto2:2] Set("SIP/6554448376-00000018", "CALLERID(num)=0426123456") in new stack |
| 17:34.44 | drenda81 | [Sep 25 19:34:09] -- Executing [s@tasto2:3] Dial("SIP/6554448376-00000018", "SIP/222") in new stack |
| 17:35.49 | drenda81 | WIMPy: but on my phone I can see only the name and NOT the number that I set.And if I recall the missed call Asterisk call the extension 222 (that is my phone) |
| 17:37.04 | drenda81 | it seems that is not possibile to manage either the name and the number, it's possible? |
| 17:43.37 | WIMPy | It surely works for me. |
| 17:45.05 | WIMPy | [TK]D-Fender: About the DIALSTATUS=CANCEL, was option d the right guess? |
| 17:47.31 | drenda81 | WIMPy: It's strage because in my cdr record I can see either the name and the number instead on my phone only the name.Can you write me the configuration of your internal phone in order to understand if the problem is in the settings of sip.conf? |
| 17:47.33 | [TK]D-Fender | drenda81: What number DO you see? |
| 17:49.48 | drenda81 | [TK]D-Fender: in my cdr record the same that I set, on my phone nothing, only the name that I set.Then if a redial this last call I can see that Asterisk is calling the extension where I make the call and not the number 0426123456 (that I set with caller id) |
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| 18:06.31 | drenda81 | [TK]D-Fender,WIMPy it's possibile that the number is overwritten by the Dial function? My dialplan is this:exten => 112,1,Set(CALLERID(name)=Name of test) |
| 18:06.31 | drenda81 | exten => 112,n,Set(CALLERID(num)=0426123456) |
| 18:06.31 | drenda81 | exten => 112,n,NoOp(Number: ${CALLERID(num)}) |
| 18:06.31 | drenda81 | exten => 112,n,Dial(SIP/223) |
| 18:06.31 | drenda81 | exten => 112,n,Hangup() |
| 18:07.00 | drenda81 | [TK]D-Fender,WIMPy maybe your are using some option in Dial() function? Thanks! |
| 18:07.36 | WIMPy | ~pb |
| 18:07.37 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 18:07.48 | WIMPy | And no, I don't use special options. |
| 18:08.51 | drenda81 | very strange,I've the same behaviour on 2 different Asterisk 1.6.xx |
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| 18:15.22 | [TK]D-Fender | drenda81: I fail to SEE the problem. |
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| 18:48.42 | drenda81 | [TK]D-Fender I try to explain with an example:If you have 2 ip phone (100 and 101) and you make a call from 100 to 101 |
| 18:49.32 | drenda81 | with this dialplan: exten => 101,1,Set(CALLERID(name)=Name of test) |
| 18:49.33 | drenda81 | exten => 101,n,Set(CALLERID(num)=0426123456) |
| 18:49.33 | drenda81 | exten => 101,n,DIAL(SIP/101) |
| 18:50.03 | drenda81 | [TK]D-Fender on your phone 101 you should be see as name "Name of test" and as number "0426123456", right? |
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| 18:50.41 | p3nguin_ | http://verydemotivational.files.wordpress.com/2010/09/7bb54014-25e9-47c1-8d64-f28470d34c96.jpg |
| 18:50.47 | drenda81 | [TK]D-Fender instead on my phone I see "Name of test" and 101 as number |
| 18:51.48 | wdoekes2 | is fromuser set? and RPID not? do a sip debug and look at the INVITE going out to 101 |
| 18:51.58 | p3nguin_ | What would you expect to see, considering you set your CID to that? |
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| 18:54.05 | Sabrina23 | Hello togerther |
| 18:54.55 | carrar | gutentag |
| 18:55.08 | drenda81 | wdoekes2: fromuser is set in either phone,rpid now is not setted in sip.conf |
| 18:55.13 | Sabrina22 | deutsche ? :) |
| 18:55.17 | carrar | no |
| 18:55.20 | carrar | thats all I know |
| 18:55.22 | Sabrina22 | :) |
| 18:55.23 | Sabrina22 | ok |
| 18:55.24 | carrar | so I let it all out :) |
| 18:55.40 | carrar | Though I did live in Berlin for a year |
| 18:55.42 | Sabrina22 | i install the Image of asterisk but cant login into the webinterface "FreePBX Administration" |
| 18:55.43 | drenda81 | p3nguin_: I'm expecting to see as name "Name of test" and as number "0426123456" |
| 18:55.54 | Sabrina22 | carrar nice |
| 18:56.14 | carrar | FreePBX isn't supported in here, try #freepbx |
| 18:56.17 | *** join/#asterisk Martin` (martin@shell.ipv6.octocore.net) |
| 18:56.17 | [TK]D-Fender | drenda81: What phone? |
| 18:56.38 | Sabrina22 | but i download it from the asterisk homepage carrar |
| 18:56.48 | Martin` | Hi there! |
| 18:56.52 | Sabrina22 | and the asterisk webinterface doenst work too |
| 18:56.54 | Sabrina22 | its offline |
| 18:56.54 | carrar | Then you didn't install FreePBX |
| 18:56.57 | wdoekes2 | drenda81: if you're looking at a SIP packet with "From: "Name of test" <sip:101@...>"" and there is no other method of CLI passing, then it's obvious that it only shows 101 |
| 18:57.07 | carrar | and there is not "FreePBX Administration" web page |
| 18:57.08 | [TK]D-Fender | Sabrina22: there is no "asterisk web interface". That is a THIRD PARTY add-on |
| 18:57.11 | Sabrina22 | do i have to start asterisk? after install the image? |
| 18:57.12 | carrar | not=no |
| 18:57.14 | [TK]D-Fender | Sabrina22: and it is NOT supported here |
| 18:57.34 | carrar | Sabrina22, personally, don't use FreePBX |
| 18:57.34 | Sabrina22 | what is asterixnow? |
| 18:57.35 | wdoekes2 | drenda81: remove the fromuser and/or add sendrpid=yes |
| 18:57.42 | carrar | Use Asterisk from source |
| 18:57.55 | Sabrina22 | not the image? |
| 18:57.55 | [TK]D-Fender | Sabrina22: A Disctro with an OS, Asterisk, and other 3rd party stuff on it |
| 18:58.01 | wdoekes2 | (the latter one only works if your phone parses RPID, many do) |
| 18:58.02 | Sabrina22 | ah ok |
| 18:58.15 | Sabrina22 | carrar without a webinterface? |
| 18:58.27 | carrar | web just slows things down |
| 18:58.31 | p3nguin_ | drenda81: No number is displayed on your phone? |
| 18:58.33 | carrar | limits you |
| 18:58.41 | WIMPy | Sabrina22: Chose between vi or emacs *eg* |
| 18:58.42 | [TK]D-Fender | Sabrina22: If you're looking for help with what you jsut installed, then you've been told where to go |
| 18:58.44 | Sabrina22 | carrar hmm from wich ressources? |
| 18:58.48 | carrar | and then you don't really learn whats going on behind the scene |
| 18:59.07 | Sabrina22 | hm |
| 18:59.24 | carrar | go here: http://www.asterisk.org/downloads |
| 18:59.46 | carrar | download 1.8 or 1.6.2 |
| 18:59.54 | carrar | and GO NUTS!! |
| 19:00.01 | [TK]D-Fender | (short trip) |
| 19:00.01 | Sabrina22 | yeah but to install and to learn its hard |
| 19:00.18 | [TK]D-Fender | Sabrina22: What do you want? |
| 19:00.25 | carrar | welcomes Sabrina22 to the good life |
| 19:01.23 | carrar | Should add that to the bot TK |
| 19:01.28 | carrar | "<Sabrina22> yeah but to install and to learn is hard" |
| 19:01.50 | drenda81 | wdoekes2: it works!!! Thanks very much!!! I forgot to remove frouser line after some old tests!! |
| 19:02.31 | wdoekes2 | np :) |
| 19:05.59 | drenda81 | wdoekes2: I can take advantage of your kindness to another small question? |
| 19:07.18 | carrar | woah |
| 19:08.16 | wdoekes2 | fire away |
| 19:09.26 | *** join/#asterisk DrDigi (~mmurphy@75-146-139-38-Stockton.hfc.comcastbusiness.net) |
| 19:09.42 | drenda81 | wdoekes2:I'm using an Italian voip provider to make calls.When I make a call to external line Asterisk report this log: |
| 19:09.53 | drenda81 | wdoekes2: [Sep 25 21:07:00] WARNING[29127]: dsp.c:1360 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 |
| 19:10.13 | p3nguin_ | And you don't know what it means, right? |
| 19:10.37 | carrar | and google is down! |
| 19:10.42 | drenda81 | wdoekes2:I'm using rfc2833 settings in ALL context. So I ask you if that line is show always |
| 19:10.51 | p3nguin_ | "Inband DTMF is not supported on codec g729. Use RFC2833" isn't clear enough. |
| 19:11.01 | drenda81 | wdoekes2: also if the settings are correct |
| 19:11.29 | drenda81 | p3nguin_ yes it's clear. But I don't understand if that warning is show only to remember us |
| 19:11.42 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:12.03 | p3nguin_ | Show something that proves you've already set it to RFC2833. |
| 19:12.04 | drenda81 | p3nguin_ that Inband dtmf is not supported on codec g729 or if it detect that I'm using inband |
| 19:12.33 | drenda81 | [general] |
| 19:12.33 | drenda81 | dtmfmode=rfc2833 |
| 19:12.37 | drenda81 | in sip.conf |
| 19:12.54 | wdoekes2 | have you tried switching codecs? |
| 19:12.58 | p3nguin_ | The warning will not appear if inband is not detected. |
| 19:13.50 | drenda81 | wdoekes2 I can use only g729 with my provider, and with internal calls (in the LAN) this line is not show |
| 19:14.29 | drenda81 | p3nguin_ it's possibile that I set correctly the rfc2833 but my provider use inband? |
| 19:16.31 | p3nguin_ | It is possible that you need to show us some SIP DEBUG already. |
| 19:18.16 | drenda81 | p3nguin_ on this Asterisk are linked several phones and the sip debug provides very long log.Is possible to catch it on a file in order to post only the relevant code? Thanks |
| 19:18.24 | wdoekes2 | drenda81: if you set faxdetect=no ? |
| 19:19.07 | p3nguin_ | Just debug for the specific IP address or peer instead of all. |
| 19:19.16 | *** join/#asterisk GameGamer43|Mac (~GameGamer@adsl-99-156-213-158.dsl.applwi.sbcglobal.net) |
| 19:20.14 | drenda81 | p3nguin_ ok |
| 19:21.25 | drenda81 | wdoekes2 wow,with faxdetect=no that line is not shown!! Why? |
| 19:21.37 | wdoekes2 | <PROTECTED> |
| 19:21.41 | wdoekes2 | <PROTECTED> |
| 19:22.12 | Martin` | Hmm is it possible to run an agi when a call is answered? :P |
| 19:22.51 | Martin` | hmm run 1 agi during a followme speech I mean |
| 19:23.23 | drenda81 | wdoekes2 uhm...ok it's clear watching the code.Now I can test if they work correctly; infact before dtmf tones was not detect during Background(file) |
| 19:24.35 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:29.03 | ChannelZ | Has the whole 'net in general been sluggish for everyone the last few days? |
| 19:29.08 | *** part/#asterisk Sabrina22 (~Sabrina@188-195-233-94-dynip.superkabel.de) |
| 19:29.45 | drenda81 | wdoekes2 also with faxdetect=no unfortunately dtmf tones are not recognized if you press a phone key during the reproduction of ivr menu (also if I'm using Background function).Instead if you press the key after ivr menu are finished it works. |
| 19:31.15 | wdoekes2 | hmyes.. faxdetect=no probably only suppresses an unrelated warning. I don't know why dtmf would fail to work while sending audio |
| 19:32.01 | drenda81 | wdoekes2 theoretically they should be work with this settings, right? |
| 19:34.45 | drenda81 | wdoekes2 maybe it's a problem of our provider because testing with internal phone, dtmf works ALWAYS! |
| 19:36.39 | wdoekes2 | I guess :) I'm out of idea's (apart from doing tedious rtp debugging, which I'm not about to do) |
| 19:38.01 | drenda81 | wdoekes2 eheh,thanks! |
| 19:46.44 | *** join/#asterisk _zoom_ (~user@41.218.41.120) |
| 19:47.13 | *** join/#asterisk fr00d (~andi@unaffiliated/fr00d) |
| 19:47.41 | _zoom_ | hi, need sip client for linux, support tls and integrated with handset |
| 19:47.52 | *** join/#asterisk Faithful (~Faithful@180.194.2.179) |
| 19:48.01 | fr00d | Hello, I'd like to setup an asterisk on my openwrt box. Can somebody point me to a really simple howto to configure asterisk so I can reach the box via a sip cient on my phone? |
| 19:48.22 | *** join/#asterisk oej (~olle@ANice-257-1-62-57.w90-52.abo.wanadoo.fr) |
| 19:59.12 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 20:02.04 | *** join/#asterisk Russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 20:02.27 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 20:02.43 | p3nguin_ | ~book |
| 20:02.43 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:02.48 | p3nguin_ | fr00d: ^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 20:04.36 | Martin` | can I create an agi thats runs in background and can follow the progress? |
| 20:04.45 | Martin` | I want to run a oommand when I pick up the phone :P |
| 20:05.09 | [TK]D-Fender | Martin`: "core show application dial" <- |
| 20:06.08 | Martin` | but it waits untill agi is completed I think? :( |
| 20:07.40 | Martin` | and I want it during a followme funtion |
| 20:07.50 | Martin` | run a command before I press 1 to accept the call |
| 20:13.45 | WIMPy | In which case(s) does DIALSTATUS get set to CANCEL? |
| 20:30.40 | ChannelZ | broken record |
| 20:39.37 | *** part/#asterisk _zoom_ (~user@41.218.41.120) |
| 20:41.08 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 20:42.03 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 21:08.59 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:09.40 | *** join/#asterisk Sabrina22 (~Sabrina@188-195-233-94-dynip.superkabel.de) |
| 21:09.43 | Sabrina22 | hello |
| 21:10.06 | Sabrina22 | can someone help me, i have final install asterisk but my telephone want connect |
| 21:10.19 | Sabrina22 | [Sep 26 00:03:33] NOTICE[1613] chan_sip.c: Registration from '"Colorhost" <sip:Colorhost@62.108.36.79>' failed for '188.195.233.94' - No matching peer found |
| 21:10.37 | p3nguin_ | No matching peer found <-- clue |
| 21:10.42 | Sabrina22 | what is that? |
| 21:10.56 | Sabrina22 | im german, cant do anything with this words |
| 21:11.00 | p3nguin_ | You are expected to have a matching peer definition in sip.conf. |
| 21:11.59 | p3nguin_ | In German: Sie vergessen, das Telefon zu konfigurieren. |
| 21:12.09 | Sabrina22 | :) |
| 21:12.12 | Sabrina22 | ah das gut |
| 21:12.40 | Sabrina22 | Ok, my telephone wants an Authname, Autentification password , username,display name |
| 21:12.41 | Sabrina22 | so |
| 21:12.45 | carrar | Can I get that in Japanese too? |
| 21:12.48 | Sabrina22 | where i find the right authname |
| 21:13.19 | carrar | ~book |
| 21:13.20 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 21:14.11 | p3nguin_ | Das telefon muss konfiguriert sip.conf. |
| 21:14.47 | WIMPy | Sabrina22: http://das-asterisk-buch.de/2.1/ |
| 21:17.01 | carrar | das blinkin lights |
| 21:17.18 | p3nguin_ | haha |
| 21:18.17 | carrar | Das machine is nicht fur fingerpoken und mittengrabben! |
| 21:18.50 | Sabrina22 | ok i think i can configure the Telephone with Extension options in PBX |
| 21:18.52 | WIMPy | What's that supposed to mean? |
| 21:19.01 | Sabrina22 | but seems like not working |
| 21:20.22 | p3nguin_ | FreePBX? |
| 21:20.33 | Sabrina22 | yeah i know the chan |
| 21:20.34 | Sabrina22 | [Sep 26 00:17:57] NOTICE[1613] chan_sip.c: Registration from '"Colorhost" <sip:1234@62.108.36.79>' failed for '188.195.233.94' - Wrong password |
| 21:20.42 | Sabrina22 | but its the right password |
| 21:21.41 | carrar | Let us know when you've downloaded Asterisk from source |
| 21:22.11 | Sabrina22 | yeah sure but i think asterisk pure is for professionals isn´t it? |
| 21:22.38 | carrar | Sounds like a catch 22 |
| 21:22.42 | WIMPy | Sabrina22: Why do you want to use Asterisk? |
| 21:23.29 | fr00d | Ok, my asterisk is "working" on my openwrt box. Can somebody tell me how to get a conceptronics c128u usb isdn adapter to work with that asterisk? |
| 21:23.51 | Sabrina22 | i habve a little company with a second girl so we want to give support for our webdesign products, but we live in different towns, we want to play a little song when someone call´s us , and rederiction if someone is not there of us |
| 21:24.08 | WIMPy | fr00d: misdn or misdn2 |
| 21:24.19 | carrar | guess you'll be learning Asterisk |
| 21:24.40 | carrar | Just think of all the new biz you will have when you can also support and install asterisk for your customers |
| 21:25.23 | Sabrina22 | hmm we are just make webdesign and php coding |
| 21:25.29 | Sabrina22 | not server operating thinks |
| 21:25.32 | carrar | Perhaps Asterisk is not for you if you do not want to take the time to learn it |
| 21:25.52 | Sabrina22 | does someone can do this professional for me so i can pay for it? |
| 21:25.58 | Sabrina22 | also set it up for money ? |
| 21:26.15 | carrar | of course |
| 21:26.25 | bougyman | Sabrina22: LOD Communications does our asterisk support |
| 21:26.31 | bougyman | they're excellent. |
| 21:26.47 | Sabrina22 | and prices? |
| 21:27.01 | bougyman | i think it's about $100something per hour |
| 21:27.02 | carrar | http://www.digium.com/en/ecosystem/ |
| 21:27.12 | bougyman | the fact that I dont' remember means they're worth it, whatever the cost. |
| 21:27.24 | *** join/#asterisk jakent (~john@ip72-205-7-182.dc.dc.cox.net) |
| 21:27.26 | bougyman | fred can do a _lot_ in an hour |
| 21:27.35 | Sabrina22 | ok |
| 21:27.38 | fr00d | WIMPy: So when I put the adapter to my ubuntu box I see a lot of stuff with hisax and hfc_usb. Is this what I need on the wrt, too, because I have misdn installed but the device is not recognized by the machine, correctly. |
| 21:27.39 | bougyman | probably your whole setup, if you had asterisk downloaded from source, would be under an hour. |
| 21:27.51 | Sabrina22 | so |
| 21:27.56 | Sabrina22 | but cant setting up it later |
| 21:28.05 | Sabrina22 | i need a running webinterface |
| 21:28.11 | bougyman | for what? |
| 21:28.13 | WIMPy | fr00d: No. hisax is I4L. That's no good for Asterisk use. |
| 21:28.17 | bougyman | it's phones, not a web app. |
| 21:28.49 | WIMPy | Sabrina22: Maybe you should consider a simpler setup? |
| 21:29.01 | WIMPy | fr00d: What Kernel do you use? |
| 21:29.14 | fr00d | 2.6.32.10 |
| 21:29.21 | Sabrina22 | [Sep 26 00:21:07] VERBOSE[1613] logger.c: -- Registered SIP '1234' at 188.195.233.94 port 5060 expires 180 [Sep 26 00:21:08] NOTICE[1613] chan_sip.c: Peer '1234' is now Reachable. (680ms / 2000ms) |
| 21:29.24 | fr00d | The newest backfire prebuild image. |
| 21:30.25 | WIMPy | fr00d: Ok, I didn't have success with misdn(1) on kernels >2.6.24, but I guess misdn2 will be too big for the box. |
| 21:30.44 | WIMPy | misdn1 needs te be added seperately. |
| 21:30.58 | fr00d | WIMPy: What do you mean with seperately? |
| 21:31.05 | fr00d | I installed the kmod-misdn package. |
| 21:31.32 | WIMPy | You need to d/l the source for bot the utils and teh kernel modules from misdn.org. |
| 21:31.49 | WIMPy | s/bot/both/ |
| 21:31.50 | fr00d | When the size will run out, I'm going to install the system onto a usb drive, so about 1 GB for this system should be enough. |
| 21:32.24 | fr00d | Uh, why do I need to download the source are the packages too old? |
| 21:32.33 | fr00d | which utils do you mean? |
| 21:32.42 | WIMPy | With misdn2 you also need LCR (linux-call-router.de). It cannot be connected directely to Asterisk. |
| 21:32.59 | WIMPy | misdnutils |
| 21:33.32 | WIMPy | And LCR is a little resource hungry. |
| 21:34.57 | fr00d | Ok, this is not that good, because ressources are scarce goods. |
| 21:35.54 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 21:35.55 | WIMPy | LCR is nice and IMHO the best solution so far, but with your hardware misdn1 might be a better choice. |
| 21:35.57 | fr00d | So I really need to build all that stuff on my own. :( |
| 21:36.18 | fr00d | yes of course, I will try misdn1 first. |
| 21:36.18 | WIMPy | I hav no experience with misdn1 and USB, however. |
| 21:36.44 | fr00d | And only if this doesn't work or does not do what I want I'll go on thining about another solution |
| 21:36.55 | *** part/#asterisk Sabrina22 (~Sabrina@188-195-233-94-dynip.superkabel.de) |
| 21:37.03 | fr00d | Hmm, damn it's getting worser and worser. |
| 21:37.19 | fr00d | I have really no experiences with building some Stuff for openwrt. :( |
| 21:37.25 | WIMPy | misdn1 has a reputation of being unstable. But in my experience it's ok if only used in TE mode. |
| 21:37.37 | fr00d | So maybe I need to get further help from the openwrt guys. |
| 21:37.52 | fr00d | What's TE Mode? |
| 21:38.04 | fr00d | Which other modes are there? |
| 21:38.06 | WIMPy | As a terminal. |
| 21:38.23 | WIMPy | As opposed to providing a network interface for terminals. |
| 21:39.10 | fr00d | Ah, ok. So is te mode what I need when I want to call through this adapter? |
| 21:39.11 | WIMPy | TE Terminal Equipment or NT Network Termination |
| 21:39.51 | WIMPy | You need TE mode to connect to a line, NT mode if you want to connect a telephone to it. |
| 21:41.25 | fr00d | I want to connect it to my telephone system, so I need te mode, right? There's no phone connected to this device. |
| 21:41.40 | WIMPy | correct then |
| 21:49.55 | fr00d | WIMPy: Thanks for your help. So I need to have a closer look to misdn. |
| 21:51.48 | *** join/#asterisk jakent_ (~john@173.6.58.158) |
| 21:52.14 | WIMPy | fr00d: Or you pay for the simple way and get another Router. |
| 21:52.14 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 21:54.00 | *** join/#asterisk jakent (~john@2001:470:8:1fc:226:8ff:fedd:93f6) |
| 21:55.41 | fr00d | WIMPy: Do you know wich one would work? |
| 21:55.54 | *** join/#asterisk jakent (~john@2001:470:8:1fc:226:8ff:fedd:93f6) |
| 21:56.00 | fr00d | But I think the router is not a big problem, but the usb isdn modul. |
| 21:56.34 | *** join/#asterisk jakent (~john@173.6.58.158) |
| 21:59.03 | WIMPy | fr00d: There is the D-Link HorstBox that comes with Asterisk, but a quite old version, or you can use a AVM FritzbBox. The hardware can't be used with Asterisk, but you can keep the origal software |
| 21:59.13 | WIMPy | along Asterisk and connect both via SIP. |
| 22:00.41 | WIMPy | FritxBoxen like e.g. 7270 come at a price, however. But it's got two ISDN ports, 1 TE and 1 NT, as well as two a/b. |
| 22:02.20 | WIMPy | Oh, and they've got DECT, that's pretty cool, if you wand cordless handsets. |
| 22:05.07 | *** join/#asterisk jakent (~john@ip72-205-7-182.dc.dc.cox.net) |
| 22:08.36 | fr00d | yeah, at my home is a fb 7270 working, but not yet on openwrt. |
| 22:19.25 | WIMPy | Not sure if you can get the original telephony application to run on openwrt. |
| 22:42.18 | drenda81 | good bye |
| 23:02.30 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 23:06.28 | *** join/#asterisk jakent (~john@2001:470:8:1fc:226:8ff:fedd:93f6) |
| 23:22.18 | *** join/#asterisk jakent_ (~john@2001:470:8:1fc:226:8ff:fedd:93f6) |
| 23:32.33 | *** join/#asterisk ectospasm (~ectospasm@user-24-236-91-170.knology.net) |
| 23:38.30 | *** join/#asterisk Sorcier_FXK (~Sorcier_F@unaffiliated/sorcierfxk) |
| 23:41.00 | *** join/#asterisk jakent (~john@2001:470:8:1fc:226:8ff:fedd:93f6) |
