| 00:00.12 | lesouvage | <PROTECTED> |
| 00:01.00 | [TK]D-Fender | ManxPower: I'm not mean about it, I just say that "marketing" is all that claim is. |
| 00:02.46 | kfife | By the way, the question earlier about app_fax: SOLUTION -- http://bugs.digium.com/view.php?id=13756 |
| 00:06.14 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 00:06.18 | lesouvage | Does one of you have problems with the latest versions of SOX? |
| 00:10.20 | baliktad | I have a provider who wants to send me calls from an entire IP range (a /24) - how can I specify the SIP peer to accept calls from a whole range of IP's? |
| 00:11.13 | *** join/#asterisk raulino (n=asdfcdf@poisson.phc.unipi.it) |
| 00:11.44 | [TK]D-Fender | baliktad: Auth'd or not? |
| 00:11.50 | raulino | hallo, i would like to register point-to-point (phone-line) analog calls and isdn bidirectional |
| 00:11.57 | raulino | is that possible with asterisk? |
| 00:12.21 | [TK]D-Fender | raulino: sure |
| 00:12.23 | raulino | which modems/isdn o voic56k should i use? |
| 00:12.35 | raulino | is there any built-in hardware stuff |
| 00:12.39 | raulino | to connect to asterisk? |
| 00:12.42 | [TK]D-Fender | raulino: Modems are not supported. Go visit the WIKI and look at the list of compatible hardware |
| 00:12.43 | [TK]D-Fender | ~wikis |
| 00:12.44 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 00:13.39 | raulino | k tnx, i would like to setup a registration server for video security and internal phone |
| 00:13.51 | raulino | is there any way to integrate everything inside asterisk? |
| 00:13.53 | raulino | plugins etc? |
| 00:14.38 | [TK]D-Fender | raulino: * is not a video surveilance app.... |
| 00:14.59 | [TK]D-Fender | raulino: if you have a SIP video capable camera perhap it'd be usable... |
| 00:15.14 | raulino | sip.. |
| 00:17.55 | raulino | tnx |
| 00:18.54 | *** join/#asterisk StephenF (n=none@198.144.201.106) |
| 00:29.17 | [TK]D-Fender | raulino: Before your next jump, you should go look at the book to get a better understanding of what * is about. |
| 00:29.19 | [TK]D-Fender | ~book |
| 00:29.19 | jbot | i guess book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 00:29.51 | raulino | i do have a basic understanding of * |
| 00:30.00 | raulino | i tought i could use the unstable video support |
| 00:30.06 | raulino | for umts video calls |
| 00:30.14 | raulino | to record analog video stream |
| 00:30.17 | raulino | someway |
| 00:30.22 | [TK]D-Fender | raulino: Nope |
| 00:30.38 | raulino | indeed |
| 00:30.50 | raulino | i found out that asterisk do support some modem |
| 00:31.02 | raulino | i need to cut hardware based costs |
| 00:32.03 | lmadsen | raulino: ummm... no, not really. And the one particular chipset it does support is not any good for production |
| 00:32.44 | raulino | i think that x100p fxo |
| 00:32.57 | lmadsen | yes, that's the one that is not any good for production |
| 00:32.57 | raulino | will do it anyway |
| 00:33.08 | lmadsen | have fun wasting lots of time |
| 00:33.13 | [TK]D-Fender | ~cheap |
| 00:33.14 | jbot | it has been said that cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 00:33.16 | [TK]D-Fender | ^^^^ |
| 00:33.44 | raulino | but it is from digium |
| 00:34.02 | raulino | isntit? |
| 00:35.21 | lmadsen | raulino: no -- it was discontinued a couple of years ago because it isn't any good |
| 00:38.09 | raulino | so lmadsen for isdns i should just look into http://www.digium.com/en/products/digital/ |
| 00:38.12 | raulino | and for digital |
| 00:38.20 | raulino | and for analog |
| 00:38.34 | raulino | http://www.digium.com/en/products/analog/ |
| 00:38.36 | raulino | here |
| 00:38.41 | lmadsen | yes |
| 00:38.47 | lmadsen | go with a product that is supported |
| 00:43.16 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:43.34 | Mark_Logan | Yeah, I mean, it's not like you are paying for base nortel server. |
| 00:44.16 | riddlebox | is there any predictive dialer software that will work with asterisk and analog lines? |
| 00:44.18 | Mark_Logan | Free is good for the system but not for the cards! |
| 00:45.58 | raulino | lmadsen: are openvox cards supported too? |
| 00:50.08 | [TK]D-Fender | raulino: they'll work, but if something goes wrong you may have issue backing up your warranty with them |
| 00:50.27 | raulino | k |
| 00:54.25 | beek | I've come to the conclusion that dealing with a PRI is a boatload easier than POTS. |
| 00:56.12 | Carlos_PHX | I would always prefer a PRI over analog. |
| 00:57.02 | beek | My problem is two POTS lines that go to my legacy PBX. My * box is in between them (FXS->PBX, FXO->PSTN). The PBX doesn't know that the line has been hung up by asterisk. |
| 00:58.01 | beek | Inserting Asterisk inline with the PRI was much easier. Today was the first day of use and it went off without a hitch. |
| 00:59.41 | lmadsen | beek: yep... all asterisk can do is guess whether the line is hung up or not. A digital circuit is significantly better |
| 01:01.15 | drmessano | Carlos_PHX: I want a multichannel analog |
| 01:01.50 | drmessano | Im gonna get one of those All Electronics Green/Red splitter kits |
| 01:07.32 | *** join/#asterisk mbranca (n=matteo@93-62-230-243.ip24.fastwebnet.it) |
| 01:07.44 | [TK]D-Fender | ~iwmwb |
| 01:07.45 | jbot | I WANT MY WEEKEND BACK! |
| 01:07.57 | [TK]D-Fender | That guy was comedy gold... |
| 01:10.26 | beek | [TK]D-Fender: Funny is when you google "What is Kewl Start" and get back this link: http://www.ashtarcommand.net/group/starfleetfederation |
| 01:12.45 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:18.38 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 01:19.19 | trnzmeta | guys: what should I google if I want to play a beep during a phone conversation |
| 01:19.30 | trnzmeta | to idicate that the phone conversation is being recorded |
| 01:20.03 | trnzmeta | currently I have my intro message state that for incoming phone calls, however external phone calls do not have that option |
| 01:21.12 | baliktad | sorry, caught up at work |
| 01:21.19 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-0a1ee16f7d26d5b9) |
| 01:21.25 | baliktad | [TK]D-Fender: this is for auth'd calls only |
| 01:27.15 | [TK]D-Fender | baliktad: baliktad then you should be able to do "host=dynamic" and use permit/deny to mask them |
| 01:29.09 | baliktad | that's what I thought as well, but it still doesn't match to the sip user |
| 01:29.48 | [TK]D-Fender | baliktad: pastebin is your friend.... |
| 01:31.19 | baliktad | ok, hang on, my boss is here |
| 01:34.03 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:36.43 | *** join/#asterisk wnspark (i=0cd8f6bd@gateway/web/ajax/mibbit.com/x-93ea74c65ec01d9a) |
| 01:37.01 | *** join/#asterisk gazzerh (n=garryh@host-212-158-227-149.dslgb.com) |
| 01:39.17 | wnspark | Has anyone in here used Teliax before? |
| 01:42.58 | [TK]D-Fender | wnspark: Several of us |
| 01:44.13 | wnspark | What is it that I need to fill out in the Teliax control panel to make the number forward to my Asterisk server? Do I need to set the destination to my server's IP address? |
| 01:44.39 | [TK]D-Fender | wnspark: No, you register to them. |
| 01:46.07 | *** join/#asterisk nikko (n=nikko@12-218-250-137.client.mchsi.com) |
| 01:48.59 | wnspark | [TK]D-Fender: What config file do I set that in? |
| 01:53.03 | *** join/#asterisk qdk_ (n=qdk@79.138.231.59.bredband.3.dk) |
| 01:56.46 | *** join/#asterisk CrazyTux (n=brandon@adsl-75-4-22-105.dsl.irvnca.sbcglobal.net) |
| 01:58.20 | [TK]D-Fender | wnspark: sip.conf... |
| 02:00.31 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 02:02.28 | *** join/#asterisk mankash (n=rom10@CPE00062575886a-CM00186832000a.cpe.net.cable.rogers.com) |
| 02:02.34 | *** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net) |
| 02:03.23 | mankash | I have simple internal setup of asterisk with 2 sip phones. |
| 02:03.49 | mankash | Is it necessary to have incomign line form outside to test voicemail etc |
| 02:03.56 | [TK]D-Fender | mankash: No. |
| 02:04.18 | [TK]D-Fender | mankash: dialplan is dialplan. doesn't matter how you choose to process your calls. |
| 02:05.09 | mankash | I have 2 sip phones. If I unregister one and then try to dial this offline one I want it to go to voicemail |
| 02:05.14 | mankash | how to do that |
| 02:07.04 | *** join/#asterisk rcy (n=rcy@d64-180-65-127.bchsia.telus.net) |
| 02:07.04 | [TK]D-Fender | mankash: Dial the exten that would call that phone and have it continue on to VM |
| 02:07.22 | mankash | it is not happening |
| 02:07.36 | mankash | ok let me try that again |
| 02:07.50 | [TK]D-Fender | mankash: Its your dialplan... did you put voicemail right after the dial? |
| 02:08.37 | mankash | sorry I don't know how to do that |
| 02:08.55 | [TK]D-Fender | mankash: Time to sit down with the book... |
| 02:08.57 | [TK]D-Fender | ~book |
| 02:08.57 | jbot | somebody said book was Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 02:09.30 | [TK]D-Fender | mankash: hree's another link for some inspiration. |
| 02:09.33 | [TK]D-Fender | ~jerjerguide |
| 02:09.33 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 02:10.06 | mankash | thx |
| 02:10.08 | [TK]D-Fender | mankash: and atiny sample for you right here : |
| 02:10.22 | [TK]D-Fender | mankash: exten => 10,1,Dial(SIP/10,20) |
| 02:10.42 | [TK]D-Fender | mankash: exten => 10,2,Voicemail(20@default,u) |
| 02:10.46 | [TK]D-Fender | mankash: there. |
| 02:10.54 | mankash | I already have 2 sip phones connetced to each other |
| 02:11.03 | mankash | measn they can call each other |
| 02:11.09 | [TK]D-Fender | mankash: You need to learn how the dialplan works. this is the absolutely most important part of * |
| 02:11.25 | [TK]D-Fender | mankash: they are not connected to each other. |
| 02:11.34 | mankash | I mean through asterisk |
| 02:11.35 | [TK]D-Fender | mankash: they are registered to ASTERISK. |
| 02:11.40 | mankash | yeah |
| 02:11.41 | [TK]D-Fender | mankash: they know NOTHING of each other |
| 02:12.19 | [TK]D-Fender | mankash: When you dial from a device like that you are dialing a number which is processed by the dialplan according to the context that your peer entry uses. |
| 02:12.58 | [TK]D-Fender | mankash: what you dial may have nothing to do with placing a call out to any other device or account. You can have 1 SIP phone and a million extensions that have nothing to do with reaching any other device |
| 02:13.09 | mankash | oh ok |
| 02:13.11 | [TK]D-Fender | mankash: You could use * as a JUKEBOX if you felt like it |
| 02:13.28 | mankash | true |
| 02:13.41 | [TK]D-Fender | mankash: Asterisk sa telephony toolkit, not a PBX. YOU are the one that can build a PBX using it if you wish |
| 02:13.53 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 02:14.37 | [TK]D-Fender | mankash: You could have NO devices of any kind in fact. You could literally run * as a jukebox right out your sound card without ANY kind of VoIP or hardware channel at all. |
| 02:14.47 | [TK]D-Fender | mankash: My * used to make me COFFEE |
| 02:15.08 | jaytee | but not lattes |
| 02:15.26 | [TK]D-Fender | (still thinking about getting it to pour, add cream & sugar, etc.... but all in good time) |
| 02:15.31 | [TK]D-Fender | ^^ |
| 02:15.38 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 02:16.09 | jaytee | we were supposed to have flying cars and Rosie the Robot by now |
| 02:16.47 | wnspark | how do I install the GUI for asterisk without subversion? |
| 02:17.03 | jaytee | by asking in the right channel for starters |
| 02:17.23 | [TK]D-Fender | jaytee: Jane, get me off this craaaa.... *wham* |
| 02:17.30 | wnspark | jaytee: is this not the asterisk channel? |
| 02:17.42 | [TK]D-Fender | wnspark: NO GUI's are supported here. |
| 02:17.49 | jaytee | [TK]D-Fender, are sprockets and cogs the same thing? |
| 02:17.49 | [TK]D-Fender | wnspark: it has its own channel |
| 02:17.57 | [TK]D-Fender | jaytee: SHHHH! |
| 02:18.17 | raulino | will i be able to record incoming phone calls from the ptsn with a asterisk configured pbx with only fxo ports? (not fxs) |
| 02:19.14 | raulino | tnx in advance |
| 02:19.14 | jaytee | raulino, yep |
| 02:19.14 | raulino | i am getting deep involved studying the asterisk pbx |
| 02:19.14 | [TK]D-Fender | raulino: EVERY call in * is just a call. |
| 02:19.21 | [TK]D-Fender | raulino: They all go through the dialplan just the same. |
| 02:19.21 | jaytee | raulino, lookup Monitor and Mixmonitor apps in "the book" |
| 02:19.39 | raulino | i need a fxs port too |
| 02:19.51 | raulino | the book? |
| 02:19.57 | jaytee | ~book |
| 02:19.57 | jbot | book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 02:19.57 | raulino | which book? |
| 02:20.29 | raulino | [TK]D-Fender: just need to get alined to english telephony terminology |
| 02:20.45 | *** join/#asterisk aces1up (n=ric@ip70-173-52-152.lv.lv.cox.net) |
| 02:21.00 | jaytee | is it me or did the pdf download link change? |
| 02:21.55 | aces1up | hey all, anyone here have a suggestion on what is better as far as quality? sangoma a200 with fxo ports or a cisco 2811 with fxo module? |
| 02:21.57 | wnspark | jaytee: it is you, if you actually go to the link provided it gives you the link to the exact same place as it was before |
| 02:22.05 | mankash | how to test this without incomig line from outside |
| 02:22.07 | mankash | exten => s,1,Answer() |
| 02:22.07 | mankash | exten => s,n,Playback(hello-world) |
| 02:22.07 | mankash | exten => s,n,Hangup() |
| 02:23.18 | jaytee | create an extension that uses Goto to jump to whatever context you put that code in and use a sip softphone or hardphone. |
| 02:24.05 | jaytee | i recall the original link just opened the book pdf in your browser |
| 02:24.12 | jaytee | and you could save it from there |
| 02:24.15 | mankash | ok |
| 02:24.46 | jaytee | I met Mark Spencer and the CEO of Digium today |
| 02:25.08 | aces1up | hey all, anyone here have a suggestion on what is better as far as quality? sangoma a200 with fxo ports or a cisco 2811 with fxo module? |
| 02:25.10 | [TK]D-Fender | mankash: dial "s" on your phone |
| 02:25.23 | jaytee | hehee |
| 02:26.19 | ManxPower | aces1up: Router based SIP Gateways just don't usually work as well as PCI cards |
| 02:26.43 | mankash | How to type s |
| 02:26.43 | ManxPower | jaytee: sorry they did not have room for me. Did ya learn much new today |
| 02:26.48 | aces1up | manxpower hrmm.. ok.. will go with the pci solution then.. why is that usually? |
| 02:27.12 | *** join/#asterisk moy (n=moy@189.169.61.171) |
| 02:27.42 | jaytee | ManxPower, the room is packed. they even had to send for more Polycom phones from the warehouse to give us by the end of the week. |
| 02:27.47 | ManxPower | aces1up: Much of the time SIP gateways can't even auth themselves, you may or may not be able to address ports as a hunt group. |
| 02:28.05 | ManxPower | Also the Cisco stuff ends up being more expensive than a PCI card. |
| 02:28.17 | jaytee | pretty much everything we covered today is basic stuff I knew already but I picked up a few things I'd missed and Jared's a good instructor. |
| 02:28.35 | aces1up | manx well we already have a 2811 with the fxo module.. |
| 02:28.51 | aces1up | but want to make sure it is reliable as we have had issues in the past with the fxo module. |
| 02:28.55 | ManxPower | aces1up: then try it and see. |
| 02:29.13 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:29.16 | ManxPower | ANALOG is really the problem. |
| 02:30.08 | aces1up | analog eh |
| 02:30.28 | aces1up | you like bandtel? |
| 02:30.37 | ManxPower | I like PRIs from the Telco. |
| 02:30.57 | ManxPower | Use VoIPoInternet for traffic spikes and vailover. |
| 02:31.01 | ManxPower | and Failover. |
| 02:31.21 | [TK]D-Fender | mankash: what are you calling from? |
| 02:34.54 | jaytee | brb |
| 02:36.16 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 02:42.10 | mankash | I tried calling s, it says service unavailable |
| 02:42.45 | [TK]D-Fender | mankash: Well is it in the a context that your phone can reach? |
| 02:44.27 | *** join/#asterisk nikko (n=nikko@12-218-250-137.client.mchsi.com) |
| 03:02.59 | *** join/#asterisk osiris (n=osiris@c-71-205-27-88.hsd1.mi.comcast.net) |
| 03:04.35 | raulino | is there any doc about difference on nt1 nt1 plus t1 e1 (digital i guess) and fxs fxo (analog), bri pri? cards? |
| 03:04.40 | raulino | i know the bare base |
| 03:05.27 | raulino | tnx in advance |
| 03:05.58 | [TK]D-Fender | ~101 |
| 03:05.59 | jbot | extra, extra, read all about it, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 03:06.09 | [TK]D-Fender | ~e1 |
| 03:06.10 | jbot | [~E1] E1 is the basic digital telephony circuit used everywhere except the US,Japan,Taiwan & Hong Kong (who use T1 and similar variants). E1 runs at 2.048Mbps. It can be an unstructured channel for data. It can be channelized, to provide 31 time slots of voice or data, each of 64kbps. Time slot 16 is used for D-Chan when used with PRI signalling. |
| 03:06.12 | [TK]D-Fender | ~t1 |
| 03:06.12 | jbot | [~T1] T1 is the basic digital telephony circuit used in North America. T1 runs at 1.544 Mbps. It can be an unstructured channel for data. It can be channelized, to provide 24 time slots of voice or data, each of 64kbps. Time slot 24 is used for D-Chan when used with PRI signalling. |
| 03:06.14 | [TK]D-Fender | ~j1 |
| 03:06.24 | [TK]D-Fender | ~fxofxs |
| 03:06.25 | jbot | rumour has it, fxofxs is An FXO port (red Digium module) expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port (green Digium module) expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it. |
| 03:06.32 | [TK]D-Fender | ~pri |
| 03:06.33 | jbot | extra, extra, read all about it, pri is [~pri] Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, R1T1,R2T1,R4T1, etc. |
| 03:06.36 | [TK]D-Fender | ~bri |
| 03:06.37 | jbot | bri is, like, [~bri] Basic Rate Interface - a form of ISDN that consists of 2 * 64kbit/s Bearer (B) channels and 1 * 16kbit/s signalling channel (D) |
| 03:06.49 | [TK]D-Fender | \o/ yay jbot spam! |
| 03:06.58 | raulino | lol tnx |
| 03:07.42 | [TK]D-Fender | raulino:ISDN- PRI (or just PRI) is a signalling over T1/E1/J1 |
| 03:10.23 | raulino | ~bri |
| 03:10.24 | jbot | rumour has it, bri is [~bri] Basic Rate Interface - a form of ISDN that consists of 2 * 64kbit/s Bearer (B) channels and 1 * 16kbit/s signalling channel (D) |
| 03:11.01 | raulino | i guess bri is nt1 plus? |
| 03:11.18 | raulino | 2 channel + |
| 03:11.22 | raulino | signal channel |
| 03:11.37 | [TK]D-Fender | raulino: looks like |
| 03:15.34 | mankash | what is the cheaper way of connecting my incoming pot analog line to my atserisk so that I can use my own voicemail system |
| 03:15.55 | [TK]D-Fender | mankash: Linksys SPA-3102 |
| 03:16.26 | mankash | what is the cost? |
| 03:18.26 | *** join/#asterisk italorossi (n=italoros@201.76.152.227) |
| 03:19.18 | [TK]D-Fender | mankash: depends where you buy it from |
| 03:22.40 | baliktad | [TK]D-Fender here's my config and failing call (trying to accept calls from anywhere in a /24 IP block): http://pastebin.ca/1251467 |
| 03:22.59 | MindTheGap_ | hwllo all, i do have libnewt installed but "make menuselect" dahdi-tools-complete refuses to let me choose dahdi_tool (saying it requires libnewt) on ubuntu 8.04 |
| 03:23.32 | *** join/#asterisk ManxPower (n=manxpowe@86.sub-75-201-166.myvzw.com) |
| 03:24.47 | [TK]D-Fender | baliktad: fromuser=17772883928 <-- comment out. |
| 03:24.59 | [TK]D-Fender | baliktad: Then if that failed, remove your permit/deny |
| 03:25.13 | baliktad | will try |
| 03:25.25 | [TK]D-Fender | MindTheGap_: need the devel too |
| 03:25.36 | MindTheGap_ | i do have it too |
| 03:27.06 | mankash | does anybody has configured openline4 card from voicetronix it use some vpb driver |
| 03:27.14 | Carlos_PHX | [TK]D-Fender: Hey, you've used a few ATAs right? I have a friend who wants to connect a home phone and a fax to our server. I never use ATAs. Any recommendation on one that does two lines and T.38? |
| 03:27.20 | mankash | somebody has given me for free |
| 03:27.32 | baliktad | :S no on both counts [TK]D-Fender, even after commenting out all three lines, still has same failure |
| 03:28.24 | [TK]D-Fender | mankash: Good luck getting that card to work with * |
| 03:29.33 | [TK]D-Fender | baliktad: I'm unsure for a range. Perhaps you should just let it fall to the context under [general] |
| 03:30.06 | baliktad | :( seems ugly, and I don't really want to accept calls from the internet at large |
| 03:30.21 | drmessano | SPA-2102 and 3102 do T.38 |
| 03:31.19 | [TK]D-Fender | baliktad: it'd be to a targeted exten. You could further check the origin of the call in your dialplan. |
| 03:31.20 | baliktad | course, this is the same provider that doesn't see anything wrong with routing all calls to 's' instead of the number dialed |
| 03:31.40 | [TK]D-Fender | baliktad: the fact it's calling "s" is YOUR fault actually... |
| 03:31.52 | baliktad | no, it's not, that's who they're routing it to |
| 03:32.02 | [TK]D-Fender | baliktad: And its because of you. |
| 03:32.02 | baliktad | look at the initial invite |
| 03:32.10 | [TK]D-Fender | baliktad: "s" is not some industry standard |
| 03:32.26 | [TK]D-Fender | baliktad: Yes, I see the invite, and know better. |
| 03:32.41 | [TK]D-Fender | balkYOU didn't tell them where to send the call to in your REGISTER statement |
| 03:32.42 | baliktad | do enlighten me as to what I can change to make the invite come to the number dialed |
| 03:33.25 | [TK]D-Fender | baliktad: REGISTER => user:pass@host/extentodialorifyouleavemeemptydon'tbesurprisedwhenitcomesbackassbecausethatswhat*willtelltem |
| 03:33.44 | Carlos_PHX | drmessano: You know if the 2102 can be used without the router part? IE, connect the internal port to the existing router? |
| 03:34.02 | [TK]D-Fender | unload chan_runonparameteres.so |
| 03:34.04 | baliktad | I have multiple DID's with them, so what exactly do you suggest I drop after the / |
| 03:34.15 | [TK]D-Fender | baliktad: You register each? |
| 03:34.20 | drmessano | I dont believe so, but it makes little difference |
| 03:34.37 | baliktad | I register one account, they send all DID's to it |
| 03:35.03 | [TK]D-Fender | baliktad: t: <sip:14252423819@ss.callcentric.com> <-- looks like a header you can strip |
| 03:35.05 | drmessano | I mean, if you plug the SPA-3102 or 2102 WAN port into your network and allow the WAN web access, it looks like an ATA |
| 03:35.26 | baliktad | yeah, they said the same thing "sort it out in your dialplan with the SIP to header" |
| 03:35.44 | Carlos_PHX | We've seen some weird stuff with double NAT (admittedly not recently). |
| 03:35.55 | [TK]D-Fender | baliktad: Great so now we've all told you. Time to get cracking! |
| 03:36.16 | baliktad | oh I can DO it in the dialplan |
| 03:36.27 | drmessano | Carlos_PHX its not double nat |
| 03:36.36 | drmessano | The SIP client is on the external interface |
| 03:36.39 | [TK]D-Fender | baliktad: Your provider is flakey so you have to live with it or change |
| 03:36.40 | Carlos_PHX | Ah, so the ATA part is |
| 03:36.42 | Carlos_PHX | Right...thanks |
| 03:36.47 | baliktad | I just don't want to. Every other provider manages to form their request properly |
| 03:36.51 | Carlos_PHX | Didn't consider that might be the case. |
| 03:36.57 | drmessano | yep |
| 03:36.59 | [TK]D-Fender | baliktad: I've encountered them before... |
| 03:37.01 | Carlos_PHX | Easy |
| 03:37.05 | drmessano | Absolutely |
| 03:37.19 | baliktad | them as in Callcentric? or them as in substandard providers |
| 03:37.21 | [TK]D-Fender | baliktad: So they are clearly the odd-ball. You know your choices.... |
| 03:37.27 | [TK]D-Fender | baliktad: YES :p |
| 03:37.30 | Carlos_PHX | He's my guinea pig, before we start doing T.38 to actual fax machines. |
| 03:37.41 | [TK]D-Fender | (on both counts, exclusive & INCLUSIVE) |
| 03:37.50 | baliktad | I really love VoicePulse |
| 03:37.58 | [TK]D-Fender | cALLCENTRIC = WINGNUTS |
| 03:38.34 | [TK]D-Fender | baliktad: VP is very friendly and open... just not so reliable |
| 03:38.41 | baliktad | but VP randomly generates DTMF tones on certain (usually female) voice tones |
| 03:39.24 | baliktad | I've found them to be reliable enough, although I did get burned cause I was using their rate API in my dialplan, and one day they stopped answering queries |
| 03:39.25 | Carlos_PHX | Heh, we had that problem years ago on old Digium cards. |
| 03:39.52 | [TK]D-Fender | baliktad: Yeah, overall I can still say they're OK, but others have had a bitch of a time it seems. |
| 03:40.40 | *** join/#asterisk Toshibi (n=ben@adsl-065-081-067-036.sip.ilm.bellsouth.net) |
| 03:40.54 | Toshibi | Hello |
| 03:41.01 | Toshibi | I have been thrown to the lions once again at work...mostly because I am always opening my big mouth about the advantages of open source. I have now been tasked with figuring out the ins and outs of configuring our medium sized (I guess) business with a PBX (Somewhere in the range of 25 to 50 phones...while maintaining our Avaya analog phones for as little cost up front as possible. |
| 03:41.08 | baliktad | I dunno how they do it, but VP still has $0.005/min to a good chunk of the US, when most everyone else is in the $0.01 - 0.02 range |
| 03:41.50 | [TK]D-Fender | Toshibi: You SURE they're analog? run them at home? |
| 03:42.32 | baliktad | Avaya: We provide a reasonable* solution at 10 times the price you can pay (*where we define reasonable) |
| 03:42.37 | Toshibi | Fender: Yes, they are sort of old but the owners still consider them a large capital expenditure. our company just grew faster than it's infrastructure |
| 03:43.07 | [TK]D-Fender | Toshibi: The cost of maintaining old phones isn't that much less than getting new phones... |
| 03:43.19 | [TK]D-Fender | Toshibi: And you lose functionailty |
| 03:43.38 | Toshibi | Fender: After looking at the converter cards, I agree. |
| 03:43.51 | [TK]D-Fender | Toshibi: Hoever what you'd look for : AudioCodes MP-124 - 24port FXS gateway. |
| 03:44.01 | Carlos_PHX | Often the old phones cost more. |
| 03:44.50 | [TK]D-Fender | Toshibi: And for smaller leftovers or if you're feel masochistic : Linsys SPA-8000 - 8port gateway |
| 03:45.05 | [TK]D-Fender | Toshibi: www.telephonydepot.com <- good sample pricing |
| 03:45.19 | [TK]D-Fender | Toshibi: that accounts for your PHONES. then comes the question of PSTN access |
| 03:45.24 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 03:45.40 | Carlos_PHX | [TK]D-Fender: Of course, he wants an ITSP... |
| 03:45.42 | Toshibi | Fender: Yes, that's the route that I had looked over. Fact of the matter is, I have to make a presentation...cost/feature/benefit/way over my head stuff....I'm just a techy |
| 03:45.42 | Carlos_PHX | :-p |
| 03:46.06 | [TK]D-Fender | Toshibi: http://www.telephonydepot.com/product_p/105-066-124-fxs.htm |
| 03:46.42 | Carlos_PHX | Toshibi: The benefits of an IP phone are quite numerous, too many to go through the list without knowing the business. |
| 03:46.50 | [TK]D-Fender | Toshibi: if you'er a techie you can already do the math for cost, you should already know the benifits & features. So really.... the only setback is you. |
| 03:46.54 | Carlos_PHX | I would go out and ask users what their annoyances are with processing calls. |
| 03:46.58 | Carlos_PHX | Then build a hit list from that. |
| 03:47.14 | Carlos_PHX | "If my phone would ___ then I could save time" |
| 03:47.50 | Toshibi | Fender: We are doing VoiP in from the Cable Co. and we want to spread that around with Standard analog phones. I have talked up Asterisk, and my familiarity with GNU/Linux would help ease the transition |
| 03:48.10 | Toshibi | Fender: That's the truth. |
| 03:48.44 | x86 | hmm... in 1.4.22 do I have to use DAHDI? |
| 03:48.50 | x86 | or is zaptel still available? |
| 03:49.11 | Carlos_PHX | VoIP in...delivered as IP or PSTN... |
| 03:49.23 | Carlos_PHX | The cable companies here turn it back into PSTN |
| 03:49.24 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 03:50.26 | Toshibi | Carlos: That's an excellent question. Another tech is dealing with the Cable Co. |
| 03:50.40 | *** join/#asterisk beastie050 (n=nick@pool-71-165-109-38.lsanca.fios.verizon.net) |
| 03:50.49 | *** join/#asterisk magenbrot (n=magenbro@ov.odn.de) |
| 03:51.04 | [TK]D-Fender | Toshibi: http://www.voipsupply.com/linksys-spa8000-g1 |
| 03:51.09 | Toshibi | Carlos: We are attempting to work together on this issue....but the other guy just has an in with the Cable people, but is otherwise brick dumb |
| 03:51.16 | beastie050 | i just installed asterisk-gui v2, and i am getting a bunch of not found errors. like after i login, it says system statys "not found" |
| 03:51.19 | beastie050 | any tips? |
| 03:51.30 | x86 | [TK]D-Fender: i only need ztdummy... do i have to use DAHDI with 1.4.22 or can I still use chan_zap? |
| 03:51.44 | [TK]D-Fender | x86: I believe you can use zap |
| 03:51.49 | Toshibi | Fender: Many more tabs and Firefox is going to implode under it's own weight. |
| 03:52.17 | [TK]D-Fender | ~nmp |
| 03:52.25 | [TK]D-Fender | hrm.. oh well |
| 03:52.35 | [TK]D-Fender | Toshibi: Your choice in opening them. |
| 03:52.46 | Toshibi | Fender: I'm just kidding around. Thanks for the help. |
| 03:52.51 | [TK]D-Fender | Toshibi: :p |
| 03:53.14 | [TK]D-Fender | Toshibi: So I'll play "bad cop" : What don't you like about what you have? |
| 03:54.16 | Toshibi | Fender: Anyhow, I have an older computer on my work bench that I'm going to demo for my bosses tomorrow...just show them the features of Asterisk from a live environ...what I use for the demo will probably not be the final production setup (I know it wont) but I would like to have some facts/figures handy. |
| 03:55.35 | Toshibi | Fender: What the company as a whole doesn't like is the fact that we've tripled in size over the last year and have 4 branch offices up and down the East Coast and one in the Midwest now, and we're not scaling while our phone bill grows astronomically |
| 03:55.54 | Carlos_PHX | Damn, just realized I have no analog phones anywhere. Can't even recall the last time I owned one. |
| 03:55.58 | [TK]D-Fender | Toshibi: And how does * change your phone bill in ways your current system can't? |
| 03:56.33 | [TK]D-Fender | Carlos_PHX: Beyond cell, what have you been running on, and how long? |
| 03:56.37 | Carlos_PHX | Toshibi: It cracks me up that people have that problem and won't spring some money for proper consulting and systems. |
| 03:56.59 | Toshibi | Fender: It's not so much about the phone bill as it is about man hours; right now, it's all human operated...we have a real life receptionist (roughly 18,000 a year) it's unprofessional, and we are starting to get busy signal complaints |
| 03:57.03 | Carlos_PHX | Doing it right the first time pays back forever. |
| 03:57.14 | Toshibi | Carlos: I know....but I work for tightwads |
| 03:57.25 | Carlos_PHX | Right, so you'd think they'd want to save money... |
| 03:57.31 | Toshibi | You'd think |
| 03:57.39 | Carlos_PHX | [TK]D-Fender: What do you mean, for my personal net connectivity? |
| 03:57.44 | [TK]D-Fender | Toshibi: So no auto-attendant. great reason to start considering a change. |
| 03:57.56 | [TK]D-Fender | Carlos_PHX: in place of "analog phone" |
| 03:58.18 | Toshibi | Fender: I agree. Consider that we have Billing, Tech Support, Sales...just in our main branch... |
| 03:58.21 | Carlos_PHX | Oh, IP phones, at home, office, on our boat... |
| 03:58.29 | [TK]D-Fender | Toshibi: My head office is ass-backwards insisting on using their receptionist AFTER having upgraded to a new Avaya IP Office |
| 03:58.38 | [TK]D-Fender | Carlos_PHX: How many years? |
| 03:58.42 | Carlos_PHX | I found a pile of PAP2s laying around, was going to play, and realized I have no phones. |
| 03:58.53 | Carlos_PHX | Hmmm...five, six years I guess? |
| 03:59.01 | Toshibi | Fender: The owners at our company like having a live person answer the phone. |
| 03:59.03 | Carlos_PHX | And cell-only for another five before. |
| 03:59.25 | Carlos_PHX | Toshibi: The vast majority of our customers do 3.5 rings to live people, then AAX. |
| 03:59.28 | Carlos_PHX | Best of both. |
| 03:59.39 | [TK]D-Fender | Carlos_PHX: Pretty impressive time frame. I didtched my analog line about 2.5 years ago |
| 03:59.39 | Carlos_PHX | If the receptionist is available, you get the human. |
| 04:00.04 | Carlos_PHX | [TK]D-Fender: I've lived in this house for 4.5 years and it has never had phone service. |
| 04:00.06 | Toshibi | Fender: But we just got word today that we are going to become major distributors of some fairly expnsive software in the south east...with a possibility of 60,000 clients |
| 04:00.10 | [TK]D-Fender | Carlos_PHX: Reason I had the line before was for DSL |
| 04:00.21 | Toshibi | Fender: Which means, we have got to upgrade (and get me some help!) |
| 04:00.24 | Carlos_PHX | Ah, we're lucky with damn-fast cable. |
| 04:00.32 | [TK]D-Fender | Toshibi: Yup... always flooding receptionists just doesn't scale |
| 04:01.00 | [TK]D-Fender | Carlos_PHX: Oh Cable was an option, but they didn't offer fixed IP's & blocked ports, etc |
| 04:01.02 | Carlos_PHX | Exactly. Most of our installs have human, then AAX, then a queue for a group of humans if they hit 0. |
| 04:01.24 | [TK]D-Fender | Carlos_PHX: basically over-contractualized fuck-offs.... with nasty early termination terms too |
| 04:01.39 | Carlos_PHX | Yeah, our cable is dynamic but we seem to get 4 months on an IP. And since we have colo space...just VPN out to that. |
| 04:01.40 | [TK]D-Fender | Carlos_PHX: Sane setup |
| 04:01.41 | Toshibi | Fender: Not at all. I really do want a system that scales. The problem is, I'm on the ground floor of a company that could become seriously....important. I am working to put myself in a place where I can get in on top of say...software/hardware infrastructure |
| 04:02.19 | [TK]D-Fender | Toshibi: Then you need to sell the presentation more than all the details. |
| 04:02.21 | Carlos_PHX | Toshibi: From that perspective... |
| 04:02.25 | Carlos_PHX | Exactly |
| 04:02.45 | Toshibi | Fender: I agree. Scalability/Professionalism |
| 04:02.45 | Carlos_PHX | And you should really work on educating yourself at a high level on all the PSTN technologies, options, etc. |
| 04:02.45 | [TK]D-Fender | Toshibi: show them the impact of what you're doing vs where you're going and "hint" at the pieces that let * scale better |
| 04:04.45 | Toshibi | Okay, I know the three big Q's they will ask me: Total Cost of Ownership, Scalability, and if it can be set up to route faxes to desktops |
| 04:05.12 | drmessano | Yes, Yes, Maybe |
| 04:05.24 | drmessano | Answer just like that |
| 04:05.58 | Toshibi | Yes is not a dollar figure :P |
| 04:06.09 | Carlos_PHX | Those are easy. |
| 04:06.13 | Carlos_PHX | Fax to desktop, yes, easy. |
| 04:06.19 | Carlos_PHX | Scale...huge |
| 04:06.24 | Toshibi | Cool, from what i have read I figured as much |
| 04:06.38 | Carlos_PHX | TCO...well, you do have time to learn, and some equipment, but no licenses, no bullshit when you want to build a new feature. |
| 04:06.39 | Toshibi | Also figured scalability was easy |
| 04:06.47 | Carlos_PHX | Self-destiny and control of the system means a lot. |
| 04:06.54 | Toshibi | I completely agree |
| 04:07.09 | Toshibi | The thing is, these guys always want plug and play |
| 04:07.17 | Toshibi | A bunch of Windows drones to be sure. |
| 04:07.22 | Carlos_PHX | As a measure of scale, our main switch runs around 700 registrations on a Xeon 2.8 |
| 04:07.24 | Toshibi | Fire and forget perhaps... |
| 04:07.25 | Carlos_PHX | And 2GB |
| 04:07.40 | Carlos_PHX | Well, plug and play = $$$ |
| 04:07.45 | Toshibi | Of course |
| 04:07.58 | Toshibi | Turn key is nice...but it's damn expensive |
| 04:08.06 | Carlos_PHX | The closest to plug and play with reasonable cost is either Switchvox or Druid. |
| 04:08.17 | Toshibi | I took a look at Druid |
| 04:08.32 | Carlos_PHX | BUT...Asterisk is like crack. You get a hit of the real thing and you can't go back to a GUI |
| 04:08.46 | Toshibi | (I've only been working on this little idea since a meeting we had at 11:30 this monring) |
| 04:09.05 | Carlos_PHX | Good, you're aggressive, that helps. |
| 04:09.17 | Toshibi | OCD is more like it |
| 04:09.26 | Carlos_PHX | Heh, well, that helps too. |
| 04:09.29 | Toshibi | lol |
| 04:09.49 | Carlos_PHX | And to wrap it up, pop in here in the am for the political talk over coffee. |
| 04:09.53 | x86 | can i use patterns in hints? |
| 04:09.53 | beastie050 | i just installed asterisk-gui v2, and i am getting a bunch of not found errors. like after i login, it says system statys "not found" anyone had that problem? |
| 04:10.01 | [TK]D-Fender | x86: not in 1.4 |
| 04:10.13 | Toshibi | I figured Asterisk would be the way to go...I know it's industry leading....I know it's what we need.... |
| 04:10.15 | [TK]D-Fender | x86: 1.6.1 will... not sure on 1.6.0 |
| 04:10.20 | x86 | exten => _7XXX,hint,SIP/${EXTEN}, etc |
| 04:10.27 | x86 | ah weak |
| 04:10.32 | [TK]D-Fender | x86: yes, I got that... |
| 04:10.38 | Toshibi | You don't want me in the middle of political talk! I'm an Anarcho-Capitalist....I just confuse issues |
| 04:10.46 | Carlos_PHX | Me too. |
| 04:11.09 | Carlos_PHX | Toshibi: What's your Linux knowledge like? |
| 04:11.27 | Carlos_PHX | Anarcho-capitalst talk radio: http://freetalklive.com/ |
| 04:12.00 | Toshibi | 4 years with Ubuntu, Knoppix before that, a little RH, a little DSL, I'm all over the place...compared to most people I know, I'm like a guru...but that isn't ahrd in these parts |
| 04:12.22 | Carlos_PHX | Oh, that's great to hear, so many people start with Asterisk with nothing on the Linux side (including me). |
| 04:12.36 | Carlos_PHX | Asterisk was my reason to use Linux. |
| 04:12.52 | Toshibi | I buy devices that I know will run linux...my MP3 player has been hacked to run Linux |
| 04:13.15 | Carlos_PHX | Well, I use an iPhone, it's a *nix at least. |
| 04:13.22 | Toshibi | Yeah |
| 04:13.42 | Toshibi | Unfortunately my day job is Windows Support, mostly.... |
| 04:14.17 | Toshibi | Though we played a dirty trick on one of our clients...they had a tablet that they kept fubaring...we put Ubuntu on it, ran the program they needed in Wine, and left it with them |
| 04:14.27 | Toshibi | Service calls from them dropped 90% |
| 04:14.28 | Carlos_PHX | Happily, my day job now is mostly Asterisk and VMware. |
| 04:15.49 | Toshibi | The guy in the office next door to me is constantly installing different Linux versions...I got him hooked and now he's as abd as I am |
| 04:15.59 | Toshibi | He swears by VMware |
| 04:16.12 | Toshibi | I used it for 6 months in Windows before I switched to nothing but Linux at home |
| 04:16.46 | Carlos_PHX | We use the enterprise stuff for servers, our own infrastructure plus deploy for clients. |
| 04:17.02 | Carlos_PHX | We do Asterisk and VMware, odd mix I guess, but two technologies we all love. |
| 04:17.12 | Toshibi | You use what works |
| 04:17.23 | Toshibi | I pick on Windows, but for most people, it works |
| 04:18.37 | jaytee | Windows is....."quaint" |
| 04:19.03 | Toshibi | Nice wording |
| 04:19.45 | hardwire | transparent |
| 04:19.56 | hardwire | lots of birds kill themselves on Windows needlessly |
| 04:20.12 | hardwire | People walk into it when it's clean.. and prefer it to be dirty so they know it's there. |
| 04:20.22 | Toshibi | Transperancy is not a word I would use for Windows....you would need a ton of Windex |
| 04:20.35 | [TK]D-Fender | hardwire: Not transparent... just full of tiny security holes whose angle of view is perceived as transparent ;) |
| 04:21.15 | Toshibi | Fender: Perhaps they should change their brand from Windows to Bug Net |
| 04:21.32 | jaytee | speaking of Windex, the Holiday Inn I'm staying at has a 32" LCD flat panel tv and you can tell the maid has used Windex or Glass Plus to clean it. Nice streaking effect going on. |
| 04:21.37 | hardwire | Toshibi: storm door |
| 04:21.51 | hardwire | Toshibi: laundry room vent |
| 04:22.03 | Toshibi | Fender: There you go....I was angling at that but couldn't think of the words....perhaps more sleep is in order |
| 04:22.41 | jaytee | if OS vulnerabilities were swiss cheese then Windows would be Alsace Lorraine |
| 04:24.15 | [TK]D-Fender | http://www.youtube.com/watch?v=jOh6Nh8w6f8 <---- :D |
| 04:24.19 | Toshibi | Actually, sleep does sound nice. 5:30 comes so so early |
| 04:25.14 | *** join/#asterisk wnspark (i=0cd8f6bd@gateway/web/ajax/mibbit.com/x-29c30fc5efa2f83c) |
| 04:25.34 | wnspark | Does anyone know of a company/person I can hire to write my config files for me for asterisk? |
| 04:25.51 | Carlos_PHX | Sleep is over-rated. |
| 04:26.05 | Carlos_PHX | Sure, anyone here can be hired to write config files. |
| 04:26.18 | jaytee | hahahahaa, "Bataan Death March that is Windows" |
| 04:26.43 | Toshibi | That was the best video I have ever seen |
| 04:26.46 | Toshibi | On youtube |
| 04:27.18 | wnspark | I really just need my basic setup to work, then from there I should be able to code the rest.... I am trying to get my Teliax number to ring to my server and for it to play back a sample sound to make sure it is working. |
| 04:27.24 | jaytee | heheheehee, a 3 hour version of Born Free |
| 04:27.34 | Toshibi | wnspark, I'll do it for the low low price of $100 per line...and get it back to you in 6 months :D |
| 04:28.02 | Toshibi | I'm just kidding around... |
| 04:28.08 | Toshibi | I may be where you are in 6 weeks |
| 04:28.24 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 04:28.31 | wnspark | Toshibi: I am pretty sure that for $30 and a few hours of my time I can read the asterisk book |
| 04:29.04 | [TK]D-Fender | wnspark: Don't worry... the betting pool on which avenue you'll actually take is going to open shortly. |
| 04:29.04 | drmessano | It only takes the book and like 3 hours of time to learn Asterisk |
| 04:29.07 | drmessano | So I hear |
| 04:29.08 | Toshibi | wnspark: That was my plan. Just trying to make an exorbitant profit |
| 04:29.28 | [TK]D-Fender | wnspark: Plenty of config samples out there, including their own site. |
| 04:29.54 | wnspark | I thought I understood how it worked, I read like 200 pages of the book and tried to follow the tutorials but it just didnt want to work |
| 04:30.06 | drmessano | Here is what you do |
| 04:30.06 | wnspark | (which means I probably did something wrong) |
| 04:30.31 | Carlos_PHX | That's probably untrue, unless by probably you mean certainly. |
| 04:31.26 | drmessano | Spend a month half-assedly learning asterisk, blame Digium+Asterisk+Dog for asterisk being buggy/shitty/too_much_cli, download trixbox, install and sell a few under the premise "you've been around asterisk for a while now" (since you've been telling yourself that anyway), and then $$$ Profit |
| 04:31.48 | drmessano | Rinse and repear |
| 04:31.49 | drmessano | Rinse and repeat |
| 04:34.25 | *** join/#asterisk terracon (n=garry@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 04:35.08 | drmessano | Then for the advanced.. Go to Astricon, since you're now an ex-spert, put yourself out there as a developer/integrator because you (1) set up asterisk to send voicemail via SMTP to %some windows server%, and (2) because you editied the resolv.conf on your trixbox. Get a pic of yourself with Mark Spencer asking for his autograph, then troll message boards for years prefixing all your statement with "when I spoke to Mark S. at Astricon.. |
| 04:35.41 | jeff | chuckles |
| 04:36.12 | drmessano | For the record, I have never spoken to Mark Spencer.. You can call it a "court order" if you like.. I like to think of it as a "legal suggestion" |
| 04:36.20 | Carlos_PHX | is trying to remember who at the Trixbox booth was doing that. |
| 04:36.39 | jaytee | drmessano, really? |
| 04:36.41 | drmessano | Oh |
| 04:36.53 | drmessano | and for a BONUS 20 points |
| 04:37.02 | Carlos_PHX | has been out with Mark, but knew fuck-all about Asterisk then. |
| 04:37.51 | jaytee | I met him today. He seemed like a nice guy but mostly talked about local restaurants. |
| 04:38.04 | Toshibi | Well thanks folks....you've been a great help (Love the open source community)...I will most certainly be back....Carlos, Fender...you guys rock! |
| 04:38.06 | Carlos_PHX | Haha, well, that was much of our conversation. |
| 04:38.16 | drmessano | Pick a developer other than Russell to claim to be "buddy of yours" after you accidentally puked on him at some Astricon afterparty, and name drop him forever, if only because it's more believable than the years you spent racing kit cars with Marko |
| 04:38.41 | Carlos_PHX | Toshibi: See you around. You have a good project ahead of you. It's how I got my start in Asterisk pretty much. |
| 04:39.06 | *** part/#asterisk Toshibi (n=ben@adsl-065-081-067-036.sip.ilm.bellsouth.net) |
| 04:39.12 | Carlos_PHX | I've been drinking with Kevin, but never puked on him. Whew. |
| 04:39.26 | jaytee | Kevin Fleming? |
| 04:39.40 | Carlos_PHX | Yeah. He founded the company I now run. |
| 04:39.59 | jaytee | he stopped in the class today and spoke a little. |
| 04:40.18 | Carlos_PHX | Ah, cool. Great guy. I've spent MANY long hours fixing stuff with him. |
| 04:40.52 | Carlos_PHX | I've also had the amusement of supporting his old systems, which run "Kevin's Linux." He also was a founder of the Linux From Scratch org. |
| 04:41.06 | drmessano | I knew Jason Parker before he was Jason Parker.. yep.. theres my name drop |
| 04:41.10 | Carlos_PHX | jaytee: So you're at Digium this week? |
| 04:41.15 | jaytee | yeah |
| 04:41.34 | Carlos_PHX | Cool. Make sure they take you to Beuregard's. |
| 04:41.37 | Carlos_PHX | And get the atomic wings. |
| 04:41.42 | Carlos_PHX | Trust me, they are mild. |
| 04:41.46 | Carlos_PHX | <snicker> |
| 04:42.10 | Carlos_PHX | Who is your instructor? |
| 04:42.20 | jaytee | I don't want mild, I want people saying, "What's that noise?" "Oh, that's just my colon screaming" |
| 04:42.49 | jaytee | Jared |
| 04:42.49 | Carlos_PHX | Kevin brought me back to Digium and walked me around saying, "This guy ate the whole atomic plate!" |
| 04:42.56 | Carlos_PHX | Ah, good speaker. |
| 04:43.02 | [TK]D-Fender | drmessano: You aren't supposed to talk about that... the WPP will snuff you out! |
| 04:43.13 | Carlos_PHX | Willing to find out stuff he doesn't know instead of bluffing. |
| 04:45.59 | drmessano | jaytee: When you get in to Digium tomorrow, check behind the Crystal Pepsi bottle in the back of the fridge in the basement and tell me if my White Chocolate Kit Kat is still there. |
| 04:47.47 | jaytee | ok, I'll check but somehow I feel like I'm being setup like Pee-Wee Herman looking for the basement of the Alamo. |
| 04:48.17 | drmessano | Send me a txt message.. I should have my beta 4G Apple iPhone on my side if I am not busy debugging Ubuntu 12 alphas with Shuttleworth. |
| 04:49.30 | jaytee | busy? how? Shuttleworth is in the Yukon saving the Polar bears |
| 04:49.46 | drmessano | Although I really need to fly to Seattle tomorrow and have lunch with Gates.. Doesn't matter how many times we vid conference a week, he's always asking me "Let's do lunch".. Wants to talk about how Ballmer is gonna screw up Windows 7 |
| 04:49.55 | *** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net) |
| 04:50.49 | drmessano | Almost makes me want to meet Branson for lunch and discuss friggin civilian spaceflight, yawn lol |
| 04:51.08 | jaytee | I've seen the vid of Ballmer doing the first commercials for Windows. That guy should be selling steak knives and vegematics on HSN |
| 04:51.10 | Carlos_PHX | I thought he was crying on your shoulder about getting hit by one of Ballmer's chairs? |
| 04:52.20 | drmessano | He does this GREAT parody of Ballmer.. we're conferencing, and he'll grab a couple bags of marshmallows out of the pantry and jump up and down yelling "Marshmallows, marshmallows, marshmallows" |
| 04:52.29 | drmessano | Its total pwnage |
| 04:52.48 | Carlos_PHX | Mmmm...marshmallows |
| 04:53.02 | jaytee | alot of people would probably love to plant hidden microphones in Ballmer's office. I'd rather plant hidden wifi speakers and just sit outside all day whispering into a microphone. "Google.............Google.............Google..................Google...............Google" |
| 04:53.23 | Carlos_PHX | Apple....Google.... |
| 04:53.25 | drmessano | HAW |
| 04:53.44 | Carlos_PHX | ....Open source... |
| 04:53.48 | drmessano | OMG that reminds me when I was at Wozniak's for thanksgiving |
| 04:54.08 | Carlos_PHX | Woz doesn't do anything for thanksgiving, so you're full of it. |
| 04:54.20 | jaytee | "CEO wanted! Must be dynamic, fat balding troll with anger management issues and clueless about technology in general." |
| 04:54.41 | drmessano | He was telling me about running into Ballmer during the summer on the coast, and was talking about the iPhone.. giving him a hard time |
| 04:55.15 | drmessano | So Ballmer asked him something about when the iPhone would support ActiveSync with Exchange.. he told Ballmer "I don't know, Google it" |
| 04:55.33 | drmessano | He said he laughed for a week afterwards.. hah |
| 04:55.55 | jaytee | if they had Ballmer's video of him sweating like a lawn sprinkler with terminal pit stains screaming at a crowd of MS employees playing on a public big screen TV in Wall Street their stock would tank overnight. |
| 04:56.15 | Carlos_PHX | Didn't their stock recently tank overnight? |
| 04:56.30 | drmessano | Oh god, what was it he said |
| 04:56.40 | drmessano | It was something stupid and cavalier |
| 04:56.44 | drmessano | Oh |
| 04:56.55 | jaytee | I hope so....my boss has a shitton of money tied up in their stock and Sun Microsystems stock |
| 04:57.07 | Carlos_PHX | So true and funny...did you see his quote on the G1? |
| 04:57.57 | drmessano | Seems like this was years ago though.. Ballmer told some reporter that tech stocks were overpriced, even Microsofts... and the price dropped 20% the next day.. He lost millions at the time. |
| 04:58.40 | Carlos_PHX | He told some reporter last week that Google is stupid, doesn't have a revenue model for the G1, and their speculation is dumb. |
| 04:58.53 | Carlos_PHX | Because, you know, Google has never made money just speculating on an idea... |
| 04:59.19 | Carlos_PHX | All Google products are carefully controlled and charged for. |
| 04:59.49 | De_Mon | hey pass me some of that coolaid |
| 04:59.54 | Carlos_PHX | If I went to my shareholder meeting, my analyst meeting, and said: 'Hey, we've just launched a new product that has no revenue model!'… I'm not sure that my investors would take that very well. But that's kind of what Google's telling their investors about Android," he said. |
| 04:59.57 | drmessano | i think it's great how Google pushed for the FCC to open up whitespace for unlicensed use |
| 05:00.13 | drmessano | Bye Bye 2.4 and 5.8GHZ |
| 05:00.15 | Carlos_PHX | Yeah, the fact that EVERYONE was against it confirms it's good. |
| 05:00.50 | [TK]D-Fender | Government increases AIG bailout to $150 billion - woohoo! |
| 05:00.58 | De_Mon | google is awesome they can take the stupidest ideas and magically make money off them without even trying. I wish I could do that |
| 05:01.04 | Carlos_PHX | Well, we can put outdoor LoS stuff on 2.4/5.8 and indoor stuff on 700 |
| 05:01.53 | drmessano | Forget those |
| 05:02.04 | drmessano | The rest of the spectrum is open now |
| 05:02.35 | Carlos_PHX | AIG... It is proof that we have become a nation of pussies. The fact that we are not at their doorstep with pitchforks and torches. |
| 05:03.21 | drmessano | You now have 54-88MHZ, 174-216, 470-512 and so on, depending on local usage |
| 05:03.39 | drmessano | 54MHZ wireless internet is gonna rock |
| 05:04.27 | jaytee | why? |
| 05:05.07 | [TK]D-Fender | drmessano: I've had so many computers 10x slower in clock than that... |
| 05:05.43 | drmessano | The Linksys WRT58zomG on some whitespace in the lower TV band will have a range 100x 2.4GHZ |
| 05:06.04 | jaytee | zomG, LOL |
| 05:07.02 | drmessano | TV is good example of government port |
| 05:07.05 | drmessano | TV is good example of government pork |
| 05:07.53 | Carlos_PHX | Mmm...pork |
| 05:07.58 | drmessano | TV was allocated some 40% of the usable spectrum between 0hz and 1GHZ |
| 05:08.07 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 05:08.09 | Carlos_PHX | wonders if it's better than government cheese |
| 05:08.17 | drmessano | Analog TV dies, and they leave this HUGE hole |
| 05:08.42 | drmessano | and the National Association of Broadcaster is still trying to hold onto it |
| 05:08.51 | drmessano | Broadcasters* |
| 05:09.51 | drmessano | Analog TV has been afforded power levels in MEGAWATTS to guarantee every rabbit eared $15 GPX TV could get a signal anywhere in America |
| 05:10.18 | drmessano | and now that Digital is redefining what is reasonable, the spectrum looks like swiss cheese |
| 05:11.04 | drmessano | The internet is helping that along too |
| 05:12.33 | drmessano | But if you take an analog TV and scan in the average market, you're get maybe 10 channels |
| 05:12.42 | drmessano | In larger markets, closer to 25 |
| 05:13.15 | drmessano | That's still 35 channels @ 6MHZ each going unused |
| 05:13.17 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 05:13.24 | drmessano | Wasted bandwidth |
| 05:13.42 | jaytee | nite all, gotta get some zzz's |
| 05:13.46 | drmessano | nite |
| 05:16.05 | Carlos_PHX | Time for some scotch, later. |
| 05:20.48 | drmessano | Stallman just called me.. need to argue some EMACS with him for a bit.. |
| 05:21.14 | orkid | stallman dont use da fone |
| 05:37.05 | *** join/#asterisk raasdnil (n=mikel@60-241-138-146.static.tpgi.com.au) |
| 05:38.02 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 05:39.07 | raasdnil | Hey guys, I have some DID lines running over an IAX2 trunk. I want to set the outbound CID depending on which extension is making the call so that the called party can dial back direct to that extension. Setting exten => s,1,SetCIDNum(<xxxxxxxxxx>) where x is the number doesn't seem to have an effect. Any ideas? |
| 05:39.38 | *** join/#asterisk mintos (n=mvaliyav@59.160.127.177) |
| 05:39.41 | raasdnil | asterisk 1.6 |
| 05:40.17 | raasdnil | I have looked around but haven't found anything definitive |
| 05:40.35 | [TK]D-Fender | raasdnil: because that app was deprecated in 1.2, and removed in 1.4 |
| 05:40.42 | raasdnil | ahh :) |
| 05:40.48 | [TK]D-Fender | raasdnil: You are folling ancient instructions that are no longer valid |
| 05:40.58 | [TK]D-Fender | following |
| 05:41.04 | raasdnil | that's why I thought I would poke my head in here and ask :) |
| 05:41.27 | carrar | Set(CALLERID(number)=8675309) |
| 05:42.02 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 05:42.03 | carrar | Set(CALLERID(name)=Jenny call me) |
| 05:43.37 | raasdnil | carrar: thanks! |
| 05:44.39 | carrar | np |
| 05:49.28 | *** join/#asterisk jakbeatz (n=jakbeatz@206.223.182.67) |
| 05:50.44 | jakbeatz | I'm having a bit of a brain fart... sip show peers should still work in 1.4.22, right? |
| 05:52.35 | [TK]D-Fender | jakbeatz: Yes |
| 05:53.14 | jakbeatz | Hmm.. So what does it mean when I get "No such command 'sip show peers' (type 'help sip show' for other possible commands)" ? |
| 05:53.54 | jakbeatz | The entire SIP command doesn't seem to be recognized on the CLI. This is 1.4.22 as part of AsteriskNOW 1.5 |
| 05:54.23 | [TK]D-Fender | jakbeatz: Go prove chan_sip is even loaded |
| 05:54.25 | raasdnil | jakbeatz: usually means that the sip module hasn't finished loading yet |
| 05:55.11 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 05:55.25 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 05:56.32 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 05:56.33 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 05:57.27 | jakbeatz | Hmm.. It doesn't seem to want to load either. Asterisk has been running for an hour or so now. modules.conf isn't configured to not load the module. |
| 05:58.10 | [TK]D-Fender | jakbeatz: what happens when you try to load it manually? |
| 05:58.35 | jakbeatz | Forgive my ignorance - I've never tried to load a module manually.. how do I? |
| 05:58.58 | [TK]D-Fender | jakbeatz: You should say that it doesn't want to then. "module load chan_sip.so" |
| 05:59.15 | [TK]D-Fender | shouldn't* |
| 05:59.53 | jakbeatz | I run module load chan_sip.so and it returns to the prompt with no error. |
| 06:00.09 | [TK]D-Fender | ok, so try again |
| 06:00.13 | [TK]D-Fender | "sip show peers" |
| 06:00.35 | [TK]D-Fender | and if you see nothing, here's a thought : "core set verbose 10" |
| 06:00.55 | lmadsen | I'm going to try and phrase this simply so I don't have to go through the whole scenario, so here goes. Assuming call recording has been enabled between the original call leg with Monitor(), how can I stop Monitor()ing a call after an attended transfer from a polycom phone? Currently if you do that, then the recording continues through to the transferred call because the original bridge was Monitor()'d |
| 06:01.23 | jakbeatz | wow... ok.. that's weird.. it's there now. How strange... why would it not load on startup? |
| 06:01.40 | [TK]D-Fender | jakbeatz: is this following my Verbose suggestion? |
| 06:01.53 | [TK]D-Fender | jakbeatz: or prior? |
| 06:03.17 | jakbeatz | Prior.. ok, I think I figured it out.. if you run safe_asterisk manually it behaves differently than if it's called from amportal.. sip module doesn't load if asterisk is started with the former command. does load if it's run with the latter command... |
| 06:03.43 | [TK]D-Fender | ok, GUI in play... I'm out... |
| 06:04.03 | jakbeatz | Well, I did say it was AsteriskNOW ;) |
| 06:04.27 | jakbeatz | Thank you for your help though anway. |
| 06:06.58 | *** join/#asterisk JohnnyBeGood (n=micko113@c-24-17-70-214.hsd1.wa.comcast.net) |
| 06:15.21 | *** join/#asterisk the_5th_wheel (n=edd@webster.cybertek.co.za) |
| 06:16.09 | *** join/#asterisk plundra (i=404@article.se) [NETSPLIT VICTIM] |
| 06:17.27 | the_5th_wheel | Hi. Can anyone reccommmend lowish cost voip gateways for use with BRI ISDN lines? |
| 06:20.51 | *** join/#asterisk LemensTS (n=matthew@adsl-70-238-180-74.dsl.stlsmo.sbcglobal.net) |
| 06:21.07 | LemensTS | <PROTECTED> |
| 06:21.07 | LemensTS | <PROTECTED> |
| 06:21.07 | LemensTS | h file or directory |
| 06:21.43 | LemensTS | the file is there, does that sound like maybe phpAGI is not there? I dont usually use agi scripts |
| 06:21.49 | drmessano | the_5th_wheel: What do you mean? |
| 06:23.15 | the_5th_wheel | Im looking for a media gateway, that isnt priced to highly, for ISDN lines |
| 06:23.39 | the_5th_wheel | At the moment i use the yenghans cards. But I want to mov away from server based isdn interfacing |
| 06:23.47 | drmessano | ah |
| 06:26.21 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 06:27.40 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 06:27.49 | *** join/#asterisk Corydon76-dig (i=two@pdpc/supporter/bronze/Corydon76-home) [NETSPLIT VICTIM] |
| 06:27.49 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 06:27.58 | *** join/#asterisk osiris (n=osiris@c-71-205-27-88.hsd1.mi.comcast.net) [NETSPLIT VICTIM] |
| 06:28.07 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 06:29.09 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) [NETSPLIT VICTIM] |
| 06:30.50 | *** join/#asterisk zr0 (i=br@unaffiliated/zr0) [NETSPLIT VICTIM] |
| 06:31.02 | *** join/#asterisk gramulhaozin (n=charles@c-76-110-248-244.hsd1.fl.comcast.net) [NETSPLIT VICTIM] |
| 06:31.25 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 06:32.18 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 06:35.36 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 06:40.36 | ManxPower | the_5th_wheel: What country? |
| 06:43.22 | ManxPower | the_5th_wheel: If you are in the USA/Canada -- you would be the last in a long line of people that tried and failed. For much of Europe, you can use the Jungans or Digium or most any other BRI card. You'll still have a server and it still will handle ISDN. |
| 06:43.43 | ManxPower | and for anything else -- well this IS the Asterisk channel. |
| 06:50.08 | LemensTS | when installing phpagi, and php5-cli, where do i put my include_path for the phpagi files? |
| 06:51.07 | LemensTS | i have wrote down /etc/php5/cli/php.ini |
| 07:02.55 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 07:05.10 | the_5th_wheel | ManxPower: My two options seem to either be getting a second server, ( i use the Jungans cards at the moment) and run that on 1.2 and then use a second server to run 1.4 for my AGI stuff. 1.4 with the bris are just giving me endless hassles where the system isnt picing up the calls always |
| 07:05.24 | *** part/#asterisk scardinal (n=supreme@90.184.100.170) |
| 07:05.55 | the_5th_wheel | How i would love it if it made financial sense to just do PRIs. But unfortunately, its much cheaper to have a wod load of BRis than a PRI |
| 07:08.47 | ManxPower | Try a PRI |
| 07:09.00 | ManxPower | Then it sucks to be you, I guess. |
| 07:09.32 | *** join/#asterisk ElSonico (n=tav@nat/ibm/x-339e7d3e39b9ae64) |
| 07:09.38 | ManxPower | You'll have your own problems if you use a media gateway. Search the mailing list archives for examples. |
| 07:18.51 | hi365 | how do i 'set things to debug mode'? |
| 07:27.49 | ManxPower | set debug X |
| 07:28.10 | ManxPower | "help" in the CLI may have provided you with that info. |
| 07:28.10 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 07:29.02 | hi365 | err, i was sure the dev that requested it was refering to something compile time... didnt think of that! |
| 07:29.06 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 07:29.23 | hi365 | whats the defualt debug level if non is sepcified? 10? |
| 07:29.54 | *** part/#asterisk raasdnil (n=mikel@60-241-138-146.static.tpgi.com.au) |
| 07:32.25 | [netman] | 0 |
| 07:32.52 | hi365 | again: if someone askes you for a debug log, what level does he want? |
| 07:33.14 | hi365 | (yes I know you dont know, but by default, what is the standard) |
| 07:33.37 | [netman] | 3 is a good level |
| 07:34.02 | [netman] | for verbose and debug |
| 07:34.51 | hi365 | any bug marshels around? |
| 07:36.02 | hi365 | please remove debug.log from #12958 |
| 07:46.50 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 07:48.20 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:48.31 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 07:49.00 | pif | hi, when building asterisk-1.4.22 I no longer get a chan_zap.so file, is that normal? |
| 07:49.49 | hi365 | seems so, as thing were moved to chan_dhadi |
| 07:49.53 | hi365 | dahdi |
| 07:50.03 | pif | but zap is still supported, no? |
| 07:50.39 | hi365 | yes it is - the chan_dahdi reades zapata.conf files, so it can really swing either way |
| 07:50.50 | *** join/#asterisk IsUp (n=nocturne@unaffiliated/isup) |
| 07:51.16 | pif | so chan_dadhi does chan_zap's job ? |
| 07:51.34 | hi365 | yup |
| 07:51.36 | IsUp | yeah, exactly |
| 07:52.34 | pif | will my zap RED alarms stay that way when "upgrading" to 1.4.22 |
| 07:52.44 | *** join/#asterisk gazzerh (n=garryh@host-212-158-227-149.dslgb.com) |
| 07:52.49 | pif | is there any other modification to do? |
| 07:53.07 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 08:10.01 | *** join/#asterisk Segnale007 (n=Pietro@host26-255-dynamic.19-79-r.retail.telecomitalia.it) |
| 08:12.26 | maqr | is anyone aware of any other pbx 'appliances' besides the trixbox one? |
| 08:15.09 | kaldemar | maqr: there are loads of them. |
| 08:17.24 | maqr | kaldemar: if i'm doing entirely IP without any t1 or analog connection, do you think i could just get away with any generic linux box? |
| 08:18.06 | kaldemar | no matter what you do, a good linux box is a good choice. |
| 08:18.16 | maqr | kaldemar: what do you think about trixbox ce? |
| 08:18.24 | kaldemar | i don't. |
| 08:18.30 | maqr | alright |
| 08:18.47 | kaldemar | i don't use any GUI to screw up asterisk. |
| 08:19.28 | kaldemar | generally speaking, people on this channel don't use GUI's. |
| 08:19.30 | maqr | yeah, me either, currently... but if i start deploying voip for people who aren't me, i think it might make sense to do something a bit easier to manage |
| 08:20.10 | IsUp | ubuntu 8.04, asterisk 1.4.. |
| 08:21.09 | maqr | IsUp: i wonder if it could be done with a livecd, a usb stick, and no harddrive? |
| 08:21.22 | kaldemar | most GUI systems are a huge pain in the ass to debug if something doesn't work, keep that in mind. i'd rather manage vanilla asterisk's. |
| 08:21.41 | drmessano | Friends dont let friends use trixbox |
| 08:23.16 | maqr | kaldemar: heh, i guess i'll boot up the image in vmware or something and see what i can make out of it |
| 08:23.30 | maqr | maybe there's all this trixbox resentment for a reason |
| 08:23.32 | drmessano | Dont use trixbox |
| 08:23.38 | drmessano | Use asterisknow beta |
| 08:23.43 | drmessano | if anything |
| 08:23.59 | drmessano | 1.5 beta has CentOS + Asterisk + FreePBX |
| 08:24.08 | drmessano | 10x better than trashbox |
| 08:24.09 | maqr | drmessano: now that looks sense |
| 08:24.11 | maqr | *sensible |
| 08:24.21 | maqr | downloads |
| 08:25.11 | SwK | isnt trixbox the same thing |
| 08:25.20 | drmessano | lol |
| 08:25.20 | drmessano | No |
| 08:25.21 | SwK | centos + asterisk + freepbx plus a few more things |
| 08:25.59 | drmessano | "plus a few more things" <--- a bunch of extra crap, including shell scripts that snoop for hardware info and report back, nevermind the broken RPMs and shoddy repo |
| 08:26.13 | drmessano | trixbox is horrid.. it's not "the same thing" |
| 08:26.24 | SwK | its all horrid imho |
| 08:26.38 | drmessano | Well thats just overgeneralizing |
| 08:26.39 | maqr | do you know if it's particularly hard to upgrade asterisknow versions? when new betas come out, is there much to do to upgrade it? |
| 08:26.49 | drmessano | yum updates |
| 08:27.06 | maqr | alright, this looks like a solution |
| 08:27.30 | maqr | drmessano: thanks, you've once again saved me from myself :p |
| 08:27.37 | drmessano | no probs |
| 08:27.49 | maqr | you'd definitely run the beta version, right? |
| 08:27.57 | drmessano | Yes |
| 08:27.59 | maqr | k |
| 08:32.05 | *** join/#asterisk CrazyTux (n=brandon@user-vcaup6e.dsl.mindspring.com) |
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| 09:00.12 | HeMan | I tried to move our conference (MeetMe) to another hardware but I get http://asterisk.pastebin.com/d653267c8 when I try to start asterisk |
| 09:00.23 | HeMan | any ideas what went wrong? |
| 09:02.16 | HeMan | I have loaded the ztdummy module |
| 09:03.11 | *** join/#asterisk ibm2 (n=Administ@196.203.192.179) |
| 09:04.26 | *** join/#asterisk mvanbaak (i=vanbaak@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 09:04.27 | HeMan | the machine is a DomU in xen |
| 09:11.57 | *** join/#asterisk ramaskez (n=RamaskeZ@212.57.232.254) |
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| 09:31.01 | *** join/#asterisk [netman] (n=netman@200.Red-88-25-139.staticIP.rima-tde.net) |
| 09:33.27 | ibm2 | hi ,can any one tell me how i can use patch h264 to asterisk1.2 |
| 09:34.29 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 09:35.45 | HeMan | is there any other way to create a conference number without using ztdummy? |
| 09:35.59 | HeMan | or perhaps without meetme |
| 09:36.01 | HeMan | ? |
| 09:39.34 | kaldemar | app_conference |
| 09:40.29 | kaldemar | but perhaps you should start by taking a look at your zaptel and zapata configuration. |
| 09:40.59 | *** join/#asterisk qdk (n=qdk@85.235.253.141) |
| 09:42.34 | HeMan | it work on our old "real" hardware |
| 09:43.11 | HeMan | I just copied /etc/asterisk to our new virtual machine |
| 09:43.33 | SwK | HeMan, you missed the zap config file thats just in /etc then |
| 09:44.43 | HeMan | SwK: umm, the old machine didn't have any /etc/zap* |
| 09:45.14 | HeMan | what to look for in the /etc/zapata.conf? |
| 09:45.23 | SwK | /etc/zaptel.conf |
| 09:46.38 | HeMan | SwK: we don't have any zaptel.conf at all on the old machine |
| 09:48.15 | *** part/#asterisk nicox (n=nicox@dsl-82-106.utaonline.at) |
| 09:48.59 | HeMan | we only have a IAX to our main asterisk and the meetme application in that asterisk config |
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| 09:49.15 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 09:50.48 | HeMan | even if I remove the /etc/asterisk/zapata.conf I get the same error |
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| 09:59.18 | *** join/#asterisk Madkiss (i=madkiss@freenode/staff-emeritus/madkiss) |
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| 09:59.19 | Madkiss | hi all |
| 09:59.36 | Shnootz | hello |
| 09:59.45 | Madkiss | just a quick question; I would like to have any incoming call from an iax2-connection piped to a sip-context. is there something to achieve this easily? |
| 09:59.59 | Madkiss | i.e. "comes in from iax2, dial exact the same extension in the from-sip context, please." |
| 10:00.15 | Shnootz | i need assistance with configuring an e1 can anyone help me on private? |
| 10:01.45 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 10:02.41 | kaldemar | Madkiss: there is no such thing as a sip-context. a context in the dialplan is a context, no matter what device uses it as it's incoming context. |
| 10:05.14 | Madkiss | kaldemar: okay, let me rephrase. I have two contexts. one is called "from-sip", it handles any sip-related stuff. one is named "from-isdn", it handles isdn-related stuff. and now i want to add a third one, "from-iax", that handles iax-related stuff. |
| 10:05.27 | Shnootz | i have a TE122 card which i'm fighting with for couple of days already can someone please assist me |
| 10:05.47 | Madkiss | kaldemar: i am well aware that this is all my local configuration design etc. pp. what i just want to achieve is that any incoming calls from the from-iax context go immediately to the from-sip context. |
| 10:06.15 | kaldemar | Madkiss: then put only include => from-sip into [from-iax]. |
| 10:07.11 | kaldemar | Shnootz: give some concrete information on your problem. |
| 10:07.59 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
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| 10:12.25 | Shnootz | i can't make the card work |
| 10:12.35 | Shnootz | the system identify it |
| 10:12.44 | Shnootz | but the calls are not coming and going |
| 10:13.01 | tzafrir_laptop | Shnootz, what card? |
| 10:13.13 | Shnootz | i made all the changes i could think of at the zapata.conf & zaptel.conf |
| 10:13.17 | Shnootz | but no luck |
| 10:13.31 | Shnootz | TE122 |
| 10:13.47 | kaldemar | by concrete information i mean an error message and pasting your configuration. |
| 10:13.51 | kaldemar | ~pb |
| 10:13.52 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 10:13.54 | tzafrir_laptop | ok. what versions of asterisk and of zaptel or dahdi? |
| 10:14.16 | tzafrir_laptop | please pastebin the output of: zaptel_hardware; lszaptel |
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| 10:19.45 | Shnootz | http://pastebin.com/m21c40b4c |
| 10:20.09 | Shnootz | anything else? |
| 10:21.20 | tzafrir_laptop | ok, so it is defined well in zaptel, and we have a problem at the asterisk configuration level |
| 10:21.33 | Shnootz | seems like it |
| 10:22.56 | Shnootz | can you point me to how it should be configured? |
| 10:24.19 | magronez | is away: cliente |
| 10:28.18 | tzafrir_laptop | what version of asterisk? What do you have in zapata.conf ? |
| 10:29.03 | Shnootz | <PROTECTED> |
| 10:30.10 | Shnootz | http://pastebin.com/m53ca259c |
| 10:30.22 | Shnootz | this is the last option i tried |
| 10:32.05 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 10:32.06 | casix | hello |
| 10:33.26 | casix | I'm installing dahdi and I have a problem. When I run dahdi_genconf it give me an error: </usr/sbin/dahdi_genconf: Cannot read '/etc/dahdi/genconf_parameters': No existe el fichero o el directorio>. I'm googling to find how to create this file but nothing found. Anyone know where can i find information? |
| 10:37.38 | Shnootz | tzafrir any ideas? |
| 10:42.28 | tzafrir_laptop | Shnootz, the first line begins with: 'channels]' Is the beginning '[' present in the original file? |
| 10:43.58 | tzafrir_laptop | casix, this is a bug that was fixed in SVN. wouraround: generate that file: touch /etc/dahdi/genconf_parameters |
| 10:45.27 | tzafrir_laptop | Shnootz, apart from that, you have three #includes . grep ^channel /etc/asterisk/zapata*.conf |
| 10:47.37 | casix | tzafrir_laptop: but it is used for dahdi_genconf to create the config files, no? how can I know what to put inside to config my card? |
| 10:48.23 | tzafrir_laptop | casix, it is an optional file . Sadly in the version you have dahdi_genconf insists that it will exist |
| 10:48.40 | tzafrir_laptop | you can find a sample for it in the source tree, under xpp |
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| 10:50.20 | casix | tzafrir_laptop: thanks |
| 10:52.11 | Shnootz | this is what i got |
| 10:52.34 | Shnootz | "" |
| 10:52.43 | Shnootz | ===/etc/asterisk/zapata-channels.conf:channel => 1-15,17-31 |
| 10:52.43 | Shnootz | ==== |
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| 10:58.59 | Shnootz | tzafrir can you logn to my system and see the conf files? |
| 11:00.44 | Shnootz | tzafrir the operator is saying that they see that the channels are blocked from our side but channel 16 |
| 11:10.05 | *** join/#asterisk IguladimNIG (n=haim@77.73.189.210) |
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| 11:17.05 | tzafrir_laptop | Shnootz, can you pastebin zapata-channels.conf ? |
| 11:17.34 | tzafrir_laptop | Shnootz, generally to ping someone, mention the full nick . Tab completion tends to help |
| 11:17.54 | Mrnick | i`m trying to get a program to work but i have not idea what this command is used for in php: <a href='irc://$ext:$ext@192.168.1.2'> can anybody help me? thank you |
| 11:21.12 | Mrnick | it`s obviously a link to an extention that has the same password on a localhost and it should make a client "VOIP ONLINE", but why the irc? |
| 11:22.31 | kaldemar | is this the right place to ask? maybe you should ask someone who wrote it. |
| 11:23.35 | Shnootz | http://pastebin.com/m46810598 |
| 11:24.01 | Mrnick | i thought someone could explain me the relation between asterisk and irc |
| 11:24.07 | Mrnick | thank you |
| 11:24.53 | kaldemar | there is none. |
| 11:25.38 | Mrnick | okay thank you |
| 11:25.55 | kaldemar | unless you make some, of course. |
| 11:26.07 | Shnootz | tzafrir my zapata-channels.conf is at http://pastebin.com/m46810598 |
| 11:28.04 | Mrnick | in that case one channel will not be enough for debugging... |
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| 11:29.18 | tzafrir_laptop | Shnootz, looking into it |
| 11:30.37 | tzafrir_laptop | the configuration looks ok |
| 11:30.47 | tzafrir_laptop | try: module unload chan_dahdi.so |
| 11:30.53 | tzafrir_laptop | module load chan_dahdi.so |
| 11:30.58 | tzafrir_laptop | (in the asterisk CLI) |
| 11:32.58 | Shnootz | as i wrote before the telephone company says that they see that all my lines are blocked but port 16, if this gives you any ideas of why it doesn't work |
| 11:33.36 | Shnootz | tzafrir_laptop, still no luck |
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| 11:34.27 | *** part/#asterisk Mrnick (n=hugo@88.197.232.70) |
| 11:35.26 | tzafrir_laptop | Shnootz, what do you see in the logs? |
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| 11:37.30 | Shnootz | tzafrir_laptop, should i configure the trunk as a group or a seperated channels? |
| 11:38.53 | tzafrir_laptop | Shnootz, what is the output of: zap show channels |
| 11:39.16 | tzafrir_laptop | in the asterisk cli |
| 11:42.08 | Shnootz | tzafrir_laptop, No such command |
| 11:42.42 | tzafrir_laptop | so chan_dahdi failed to load |
| 11:43.16 | tzafrir_laptop | now check the logs to see why it has failed |
| 11:43.18 | Shnootz | tzafrir_laptop, i did amportal restart and now it works |
| 11:44.00 | Shnootz | tzafrir_laptop, http://pastebin.com/m26017348 |
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| 12:25.35 | Shnootz | tzafrir_laptop ??? |
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| 12:34.49 | tzafrir_laptop | Shnootz, so all's well, right? |
| 12:34.55 | tzafrir_laptop | what doesn't work? |
| 12:36.51 | Shnootz | i still can make calls |
| 12:36.55 | gambler1 | Hi, is it possible to execute stored procedure after the call is ended? (to write cdr in database and calculate the new value of credit field) |
| 12:37.20 | Shnootz | tzafrir_laptop, should i treat it as indevidual channels or as a group? |
| 12:38.50 | Shnootz | on the trunk configuration |
| 12:43.42 | magronez | is back |
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| 13:01.22 | *** join/#asterisk RMod (n=nicolasj@unaffiliated/rmod) |
| 13:01.47 | IguladimNIG | tzafrir, shnootz E1 provider is using simens switch, we are using span=1,1,0,ccs,hdb3,crc4 with a te122p card, can it be timing issue? |
| 13:02.58 | IguladimNIG | like sync with the switch? we operated the same E1 with a hybrid PBX and it was working |
| 13:03.38 | tzafrir_laptop | a timing issue for what, exactly? |
| 13:04.02 | *** join/#asterisk Tuxguy (n=homeins6@ip-208-109-154-197.ip.secureserver.net) |
| 13:04.15 | Tuxguy | If I have one DID coming into my system, how many outbound lines is that? 2? |
| 13:04.29 | IguladimNIG | for the sync with the Simens switch? |
| 13:06.15 | IguladimNIG | I get masseges like D-Channel down and then UP and again but with the hybrid it was stable |
| 13:06.23 | Tuxguy | Say for example, I place a call to person b, then a call to person c, and bridge them, and i disconnect, is that still using my line? or does that become their connection? |
| 13:07.16 | tzafrir_laptop | IguladimNIG, that means you take timing from them |
| 13:07.33 | IguladimNIG | yes I am it is on PRI_CPE |
| 13:07.37 | tzafrir_laptop | IguladimNIG, how often do you get those messages? Any alarms? |
| 13:07.54 | *** join/#asterisk djin (n=djin@89.146.42.209) |
| 13:08.09 | IguladimNIG | no alarms, just this and it can come 10-15 in a row in 1-3 sec in between |
| 13:08.56 | IguladimNIG | but strangly, with the hybrid it is stable. |
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| 13:17.37 | tzafrir_laptop | IguladimNIG, resetinterval=never ? (nah, I'm probably missing something) |
| 13:17.50 | tzafrir_laptop | What do you see in 'pri intense debug span 1'? |
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| 13:20.29 | *** join/#asterisk IsUp (n=nocturne@unaffiliated/isup) |
| 13:20.37 | IsUp | hi ya |
| 13:22.20 | IsUp | i have Asterisk with Public IP, connected to my network and a grandstream connected to this network (phone 1). and i have another grandstream (phone 2), its behind the nat. when i am trying to call phone 1 to phone 2, i am getting one way audio or no audio. |
| 13:22.39 | IsUp | i've read articles on voip-info.org about SIP NAT solutions but it didnt work. |
| 13:25.51 | IguladimNIG | tzafrir, huge pile of masages, basicly none looks like a problem, I also opend the trunks as zap/1-31 insted of a group incase... |
| 13:26.47 | IguladimNIG | what is making things strange is that from the switch side (provider) they see like all the channels are blocked but channel 16 |
| 13:28.32 | IguladimNIG | supporting that is that the maseges on the regular -rvvvvv i see that when a call is comming out it is looking for a channel to call and than (after few tries on deferent channels) i get the noavil massege. |
| 13:29.08 | *** join/#asterisk shido6 (n=shido6@209.114.208.111) |
| 13:31.44 | *** join/#asterisk sosperec (n=david@office.axpnet.com) |
| 13:31.49 | sosperec | Hello! |
| 13:32.00 | sosperec | How can I use the pickup application? It works well for internal calls, but fails for the outside world |
| 13:34.06 | IsUp | IguladimNIG: did you try 'zap show channel xx'? and whats your signalling? |
| 13:38.03 | IguladimNIG | IsUp, my signaling is euroisdn |
| 13:43.57 | IguladimNIG | IsUp, in the zap show channel 1 the signaling is pri isdn |
| 13:44.18 | IsUp | okay, can you get calls? |
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| 13:47.09 | *** join/#asterisk italorossi (n=italoros@201.76.154.111.intranet.digi.com.br) |
| 13:52.09 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
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| 13:54.37 | IsUp | hey [TK]D-Fender |
| 13:57.50 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
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| 14:06.29 | Katty | morrrrning. |
| 14:10.17 | IsUp | hey ya |
| 14:13.29 | IsUp | Katty: i have trouble with one way audio or no audio. any ideas? i've tried NAT stuff. |
| 14:13.50 | Katty | sounds like rtp ports. |
| 14:14.13 | Katty | you might try having a look at your firewall log. |
| 14:14.25 | Katty | unless it's lan. |
| 14:15.09 | IsUp | Asterisk is on Public IP and not firewalled. |
| 14:15.12 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 14:18.09 | *** join/#asterisk VoipForces (n=someone@67.55.25.219) |
| 14:18.41 | VoipForces | Hi, anyone knows a softphone that supports the url parameter of the queue app ? |
| 14:26.18 | IguladimNIG | IsUp, no no calls in or out, I am installing a new box with another card, but I think is is a long shoot, no other idea. |
| 14:27.14 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 14:29.03 | Katty | [TK]D-Fender: http://www.flickr.com/photos/izaah/3022326232/ |
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| 14:37.59 | nicox | Hi |
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| 14:41.04 | [TK]D-Fender | IsUp: Then its most likely that you're allowing your phones to reinvite |
| 14:41.53 | nicox | sometimes, i get this warning in my logs: chan_iax2.c: Call rejected by 10.x.y.z: No authority found and i have no idea why |
| 14:42.18 | IsUp | [TK]D-Fender: canreinvite=no in my caller and called party |
| 14:42.34 | [TK]D-Fender | IsUp: put it EVERYWHERE |
| 14:43.22 | IsUp | okay, in sip.conf [general] context, i am setting. |
| 14:43.30 | [TK]D-Fender | IsUp: and every peer |
| 14:43.37 | IsUp | okay |
| 14:43.42 | IsUp | what about phone configuration? |
| 14:43.46 | IsUp | should i configure anything?.. |
| 14:44.49 | IsUp | umm, i did the 'canreinvite' in my sip.conf but it didnt work. |
| 14:44.50 | [TK]D-Fender | IsUp: go look at SIP debug, and feel free to show your configs, firewall, etc |
| 14:45.20 | *** join/#asterisk sdaniels (n=chatzill@216.65.195.52) |
| 14:45.21 | IsUp | okay, let me prepare. its a production server, getting debug is too difficult =) |
| 14:45.58 | magronez | is away: aloco |
| 14:47.37 | *** join/#asterisk miloux (i=milu@213.88.194.123) |
| 14:48.08 | *** join/#asterisk Spirits-Sight (n=christop@c-24-218-128-159.hsd1.ma.comcast.net) |
| 14:48.52 | tzanger | grr |
| 14:48.59 | tzanger | this stupid adit600 won't generate ring voltage on its fxs ports |
| 14:49.11 | Spirits-Sight | Can Asterisk be used with out adding any hardware and just using only VOIP |
| 14:49.23 | Shnootz | tzafrir_laptop, are you still here? |
| 14:49.48 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 14:49.53 | tzafrir_laptop | yes |
| 14:49.54 | [TK]D-Fender | Spirits-Sight: Yes. |
| 14:50.28 | Spirits-Sight | I am reading that book that was recommended yesterday to me |
| 14:50.33 | [TK]D-Fender | Spirits-Sight: And * can be use with even VoIP. You could use * as a JUKEBOX with jsut your sound card if you wanted. You could use * as a CRON replacement. Or any other number silly things |
| 14:50.47 | [TK]D-Fender | without* |
| 14:51.10 | *** join/#asterisk brodiem (n=brodiem@e2.72.1243.static.theplanet.com) |
| 14:51.24 | nicox | Hi, can somebody help me with this problkem? sometimes, i get this warning in my logs: "chan_iax2.c: Call rejected by 10.x.y.z: No authority found" and i have no idea why |
| 14:51.48 | [TK]D-Fender | nicox: because your system is getting calls with bad auth. |
| 14:51.57 | [TK]D-Fender | nicox: Go find out what is sending them |
| 14:52.21 | Spirits-Sight | [TK]D-Fender: wow, is it possible to setup a basic setup with out having to learn so much that it throws it out the window for me, right now I just want a way to replace a 60 bill to a less money as state yesterday |
| 14:52.40 | nicox | the calls are from this system |
| 14:53.04 | nicox | and much much calls go through without problems |
| 14:53.06 | [TK]D-Fender | Spirits-Sight: * has a learning curve to it. read the book, look at this guide as a sample of how simple a setup can be. |
| 14:53.08 | [TK]D-Fender | ~jerjerguide |
| 14:53.09 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 14:53.14 | nicox | but some calls are rejected |
| 14:53.43 | [TK]D-Fender | nicox: Time to really look at the ones that work and the ones that don't |
| 14:53.49 | jameswf | ~book |
| 14:53.50 | jbot | methinks book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 14:54.12 | brodiem | Anyone able to point me to the bug report for this issue, I cannot find anything on the problem but I know of at least a few installations with the problem: When using app_queue, the last member only rings a partial ring (i.e. 1-2 seconds), but the CLI indicates "nobody answered in 15000ms" or whatever actual ring time is set. |
| 14:54.17 | Spirits-Sight | [TK]D-Fender: I want to be able to make out going call and want to be able to keep unlimitied incoming calls for my 800 number |
| 14:54.31 | [TK]D-Fender | Spirits-Sight: my answer is not changing. |
| 14:55.01 | [TK]D-Fender | Spirits-Sight: Its as complicated as your needs are. |
| 14:55.19 | [TK]D-Fender | Spirits-Sight: Go look at that guide for some inspiration. |
| 14:55.38 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:56.02 | Spirits-Sight | I know, I am reading, I just want to make sure its worth the time right now to do it, is what I am asking so much in the cost area possible do you know (keeping the 800 number with unlimited incoming calls |
| 14:56.08 | *** join/#asterisk djin (n=djin@89.146.42.209) |
| 14:56.17 | nicox | the calls looks like the same, (the only thing which is different are the destination numbers.) |
| 14:56.41 | [TK]D-Fender | Spritis go look at what providers charge and see whose plan suits you |
| 14:56.44 | [TK]D-Fender | ~itsplist-us |
| 14:56.45 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 14:57.47 | nicox | on this systems there are about 100+ calls a minute, and this error happens ever 2 hour or so on |
| 14:58.17 | Spirits-Sight | [TK]D-Fender: the link is not loading for some reason |
| 14:59.19 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-186-176-190.bflony.east.verizon.net) |
| 14:59.28 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 14:59.31 | [TK]D-Fender | Spirits-Sight: http://74.125.113.104/search?q=cache:x-SVjZ-02u8J:www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/+http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/&hl=en&gl=ca&strip=1 |
| 15:02.14 | Spirits-Sight | [TK]D-Fender: one last question, is there any things I should do different for Ubuntu-Server |
| 15:02.29 | nicox | [TK]D-Fender: any idea? |
| 15:03.06 | [TK]D-Fender | Spirits-Sight: Yes. Go Google Ubuntu Asterisk isntall guides. |
| 15:03.25 | [TK]D-Fender | nicox: You haven't shown anything. There is nothing to be said for your problem. |
| 15:03.34 | Spirits-Sight | thank, do this before following the other link you gave right |
| 15:04.08 | nicox | [TK]D-Fender: what information are useful to solve a problem like this? |
| 15:04.27 | IsUp | ~paste |
| 15:04.28 | jbot | extra, extra, read all about it, paste is http://rafb.net/paste/, or see also pb |
| 15:04.30 | IsUp | ~pb |
| 15:04.31 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:04.36 | SuPrSluG | spirits, only difference is the sudo command in ubuntu |
| 15:07.17 | Spirits-Sight | Ok, I did a goolge search and the stuff I am finding also seems to be other stuff to and not just for installing asterisk one is with FreePBX and another one is with foneBRIDGE2 and so on, sorry for being a pain |
| 15:07.34 | Spirits-Sight | I want to learn it right and do it right not |
| 15:08.20 | IsUp | [TK]D-Fender: ok, i did it. here is my outputs. http://pastebin.ca/1253318 http://pastebin.ca/1253319 also draw a basic flow about my schema http://imagebin.ca/view/8iFbSmcT.html |
| 15:09.28 | IsUp | i got this debug when boy1 is trying to call boy2 then boy1 hangs up. no audio on both side. |
| 15:09.56 | IsUp | debug seems complicated, i've tried to truncate other outputs. (which not belongs to my prob) |
| 15:10.52 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 15:16.35 | *** join/#asterisk freakazoid0223 (n=mattc@pool-72-81-0-69.phlapa.east.verizon.net) |
| 15:18.27 | SuPrSluG | ru forwarding rtp ports on your router? |
| 15:20.24 | Katty | wibbles |
| 15:20.26 | Katty | wobbles |
| 15:20.56 | VoipForces | Hi, anyone knows if there is a way to execute dialplan code on a sip registration failure ? |
| 15:21.35 | [TK]D-Fender | Spirits-Sight: "the stuff". how... generic. Fonebridge is HARDWARE, and FreePBX is a bolt-on GUI interface. |
| 15:22.59 | [TK]D-Fender | IsUp: permanently remove all commented lines from your sip.conf. |
| 15:23.11 | [TK]D-Fender | IsUp: They confuse what is and is not set |
| 15:23.40 | Spirits-Sight | I didn't know what to call it "the stuff", I found that Ubuntu has asterisk in its repro, so I let it install and now I am following the setup of the link you gave, it seems throw that Ubuntu did alot of the settings for you |
| 15:24.00 | IsUp | yeah i know =) but i kept original before posting. what should i do [TK]D-Fender? |
| 15:24.04 | [TK]D-Fender | IsUp: And your local phones should be nat=no |
| 15:24.05 | Spirits-Sight | is this a good idea or not |
| 15:24.30 | [TK]D-Fender | Spirits-Sight: Go google a guide for installing * from sourse on Ubuntu |
| 15:24.47 | IsUp | what about nat=yes in [general] context? should i remove it? |
| 15:25.07 | *** join/#asterisk mocker (i=ksexton@198.247.173.227) |
| 15:25.41 | [TK]D-Fender | IsUp: if * is public, it should be "nat=no". Your remote phone should be nat=yes, qualify=yes, and EVERYWHERE you should have "canreinvite=no" |
| 15:26.09 | *** join/#asterisk tumisho (n=rael@196.41.8.89) |
| 15:26.47 | tumisho | goodday everyone |
| 15:27.27 | IsUp | [TK]D-Fender: okay setting now. |
| 15:27.58 | tumisho | I am trying to make asterisk work as a smsc for analog phones but I cannot get it right |
| 15:29.00 | IsUp | okay Fender, now i can call from boy2 to boy1 |
| 15:29.20 | IsUp | but boy1 cannot call boy2. Asterisk says UNREACHABLE for remote phone. |
| 15:29.28 | IsUp | so qualify packets are lost or something.. |
| 15:29.54 | mocker | bleh. asterisk isn't being compiled w/ zaptel support for some reason on this box. |
| 15:30.07 | IsUp | i am trying to remove qualify= and calling again. phone is ringing but no audio on both sides |
| 15:31.16 | [TK]D-Fender | IsUp: have your remote phone reregister |
| 15:31.52 | IsUp | ah no, let me restart phone with qualify=yes. |
| 15:32.38 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 15:33.03 | IsUp | ok, i see "registered" in CLI, but still UNREACHABLE. |
| 15:33.33 | mocker | compile and install libpri, check. compile and install zaptel, check. compile and install asterisk, check. So why when I do 'show channeltypes' is zap not listed? |
| 15:34.18 | IsUp | zaptel first and then libpri. |
| 15:34.39 | [TK]D-Fender | mocker: did you trash your * source folder and reextract from scratch? |
| 15:35.14 | mocker | [TK]D-Fender: Yeah, just did that. |
| 15:35.16 | mocker | Still no go. |
| 15:35.31 | [TK]D-Fender | mocker: load the module manually |
| 15:35.49 | IsUp | mocker, what about kernel headers? did you install them? |
| 15:35.52 | [TK]D-Fender | mocker: So far I also don't trust that zaptel is started prior to * |
| 15:36.27 | mocker | ztcfg -vvvv comes back fine. |
| 15:36.39 | IsUp | whats your card? |
| 15:37.30 | mocker | Wildcard TE405P quad-span |
| 15:37.48 | IsUp | okay, do 'lsmod | grep zaptel' in your shell. and paste output. |
| 15:38.00 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 15:38.41 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 15:38.56 | *** part/#asterisk Spirits-Sight (n=christop@c-24-218-128-159.hsd1.ma.comcast.net) |
| 15:40.11 | IsUp | [TK]D-Fender: what you can suggest for my prob? |
| 15:42.36 | mocker | lsmod | grep zaptel comes back w/ the modules loaded. |
| 15:43.25 | IsUp | can you use any "zap" command on CLI? |
| 15:43.38 | [TK]D-Fender | IsUp: I suggest you show me debug for your calls and your new configs because right now you seemt o think I trust them or am psychic :) |
| 15:44.08 | jameswf | [TK]D-Fender: does a shockingly good impression of madam cleo |
| 15:44.24 | IsUp | ok =p i'll post new config and debug stuff. |
| 15:45.40 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:46.05 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 15:46.35 | *** join/#asterisk kensuke_ (i=c829e4f4@gateway/web/ajax/mibbit.com/x-2db0f56f794731f3) |
| 15:46.43 | mocker | IsUp: No.. |
| 15:47.16 | mocker | sorry, people jumping in and out of my office. |
| 15:47.38 | mocker | IsUp: http://pastebin.ca/1253342 |
| 15:48.14 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
| 15:49.28 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:51.05 | kensuke_ | Hi |
| 15:51.20 | kensuke_ | how can send "sip debug" to a file? |
| 15:51.36 | orkid | log |
| 15:51.39 | orkid | :) |
| 15:51.54 | orkid | just maybe |
| 15:53.35 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 15:53.41 | kensuke_ | mm, no appear on the message log and in debug log |
| 15:54.02 | mocker | Hmm, might as well just start from scratch and recompile everything. |
| 15:54.58 | kensuke_ | :O, mmm, maybe |
| 15:55.07 | *** join/#asterisk Spirits-Sight (n=christop@c-24-218-128-159.hsd1.ma.comcast.net) |
| 15:55.17 | mocker | kensuke_: Not talking to you. :) |
| 15:55.57 | IsUp | mocker, stop your asterisk and start with: asterisk -vvvvvvvvgc |
| 15:56.07 | IsUp | probably you can see chan_zap error. |
| 15:56.17 | IsUp | and paste outputs. |
| 15:56.47 | mocker | IsUp: I'm reinstalling, but I'll try that next. |
| 15:56.49 | mocker | Thanks for the tip. |
| 15:56.55 | Spirits-Sight | under voicemail.conf -> general -> serveremail= <- I don't know how I would enter this for a email server on google, please help |
| 15:59.27 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 15:59.27 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:03.57 | *** join/#asterisk Bad_Robot- (n=BadRobot@cpe-76-173-219-25.socal.res.rr.com) |
| 16:09.19 | ManxPower | Spirits-Sight: What does voicemail.conf.sample tell you? |
| 16:09.24 | magronez | is back |
| 16:09.58 | ManxPower | I strongly doubt GMail will permit untrusted, random people from using them as an e-mail server. |
| 16:10.13 | Spirits-Sight | Well if I did it right then I can use a any email address thats vaild |
| 16:10.55 | Spirits-Sight | well, I do use them as a email server under they apps |
| 16:11.22 | ManxPower | ; Who the e-mail notification should appear to come from |
| 16:11.22 | ManxPower | serveremail=asterisk |
| 16:11.28 | ManxPower | pretty straightforward |
| 16:13.17 | Spirits-Sight | ok, I was just making sure, thats what I have |
| 16:16.33 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:16.34 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 16:17.56 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:20.04 | *** join/#asterisk knightfal (n=knightfa@66.178.134.235) |
| 16:21.17 | knightfal | Good Morning Guys |
| 16:21.58 | knightfal | Im having a wierd issues. When I try to get into the CLI I get a No more connections allowed error. as show here http://pastebin.com/m6e5ead65 |
| 16:22.04 | knightfal | Anyone have any Ideas |
| 16:22.26 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 16:23.17 | IsUp | (pid = -1) |
| 16:23.18 | Spirits-Sight | Ok, I followed the how to on jeremy-mcnamara.com on a simple setup and now the phone says fail, I believe I followed the how-to by the T |
| 16:24.14 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-dd3a4d924ba4c404) |
| 16:24.14 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:28.11 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 16:29.23 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 16:30.09 | ManxPower | Spirits-Sight: That is the problem with blindly following instructions -- You don't understand what you are doing so you don't know how to fix anything. |
| 16:30.34 | ManxPower | What is the ACTUAL error messages (use pastebin if it's more than 2 lines) |
| 16:31.24 | *** join/#asterisk masus (i=masus@88.248.14.186) |
| 16:31.46 | Spirits-Sight | please give me a tiny bit of credit, I am willing to learn its just not easy and I was told to follow the hotot and I get what the howto was saying, any how the phone only shows fail on the screen of the phone |
| 16:32.17 | *** join/#asterisk [8none1] (n=aherbert@cerberus.franklinamerican.com) |
| 16:32.38 | Spirits-Sight | here is what the CLI says: chan_sip.c:15236 handle_request_register: Registration from '"Ext 1" <sip:100@192.168.1.109>' failed for '192.168.1.104' - No matching peer found |
| 16:32.57 | ManxPower | do you have a [100] section? |
| 16:33.02 | ManxPower | in sip.conf |
| 16:33.06 | Spirits-Sight | yes |
| 16:33.25 | ManxPower | pastebin your sip.conf, masking only passwords |
| 16:34.03 | Spirits-Sight | ok |
| 16:34.08 | ManxPower | Spirits-Sight: Have you read the Asterisk Book? |
| 16:34.09 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 16:35.12 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 16:35.24 | Spirits-Sight | http://pastebin.com/m1777a439 , the passwords are fake, and I am reading and have been today |
| 16:35.41 | Spirits-Sight | the future one right |
| 16:35.53 | Spirits-Sight | 2 editon |
| 16:36.09 | *** join/#asterisk Ccomp5950 (n=Ccomp595@66.190.102.236) |
| 16:36.15 | *** join/#asterisk riddlebox (n=adfad@75-128-170-26.static.stls.mo.charter.com) |
| 16:36.39 | ManxPower | Try changing [100] to [Ext 1] do a reload and see of anything works or gives you a different error message. |
| 16:36.49 | *** join/#asterisk ifnotwhynot (n=davidh@196-209-94-84-tpr-esr-2.dynamic.isadsl.co.za) |
| 16:37.15 | ManxPower | what device is 192.168.1.104 and what device is 192.168.1.109? |
| 16:37.42 | ifnotwhynot | hi there is there a way to shorten the delay when receiving a call from zap fxo to sip extension? |
| 16:37.46 | masus | hi all, ,isit possible to capture messages that are send to asterisk manager "5038" |
| 16:37.47 | riddlebox | on zap channels(analog), as soon as a call goes out of them is it considered answered by asterisk? |
| 16:38.11 | ManxPower | masus: Yes, but not using Asterisk. You can use something like tcpdump or wireshark. |
| 16:38.14 | ifnotwhynot | it takes forever before sip extensions start to ring on starts on ring count 5 |
| 16:38.27 | masus | ManxPower: ok Thanks |
| 16:38.28 | ManxPower | riddlebox: Correct. There is no answer indication on analog lines. |
| 16:38.48 | ManxPower | ifnotwhynot: pastebin the CLI output of a problem call. |
| 16:38.51 | Spirits-Sight | is the error after the reload command: chan_sip.c:15236 handle_request_register: Registration from '"Ext 1" <sip:100@192.168.1.109>' failed for '192.168.1.104' - No matching peer found |
| 16:38.54 | ifnotwhynot | k |
| 16:39.02 | riddlebox | ManxPower, so I can only get that from a sip trunk or a pri then correct? |
| 16:39.05 | ManxPower | Spirits-Sight: and my 2nd question? |
| 16:39.27 | Spirits-Sight | sorry did not see the question let me go back |
| 16:39.30 | ManxPower | riddlebox: Well PRI or VoIP w/PRI PSTN connection. SIP. IAX, whatever protocol |
| 16:39.37 | ManxPower | (10:37:15 AM) ManxPower: what device is 192.168.1.104 and what device is 192.168.1.109? |
| 16:39.51 | ManxPower | all ITSPs would be using PRIs |
| 16:40.29 | ManxPower | riddlebox: Actually the issue is that the line/port is FXO, FXO can be transported on analog or T-1 (non-PRI) |
| 16:40.44 | riddlebox | ManxPower, now the thing is to find a provider that will send a paper bill to this church, and one that will do like two-three lines with rollover |
| 16:40.45 | Spirits-Sight | device .104 is a spa 942 (if remember right) and .109 is the system with asterisk |
| 16:41.37 | ManxPower | Spirits-Sight: factory reset the SPA. Then set ONLY proxy server, userid, password on the phone. See what happens. |
| 16:42.13 | ifnotwhynot | got it working diabled callerid and made immediate=yes working fine |
| 16:42.24 | ManxPower | ifnotwhynot: turn off immediate=yes. |
| 16:42.35 | riddlebox | how well would two or three sip/iax trunks work through DSL? |
| 16:42.38 | ManxPower | when you disable callerid, you don't need immediate=yes and it could cause issues in the future. |
| 16:42.49 | ifnotwhynot | thx |
| 16:42.51 | ManxPower | riddlebox: as well as anything using the internet. |
| 16:43.00 | ifnotwhynot | what does the emmediate do? |
| 16:43.35 | riddlebox | I wonder if I can talk them into purchasing a VoIP line and only use it for this predictive dialing |
| 16:43.55 | ManxPower | ifnotwhynot: it causes asterisk to process the call as soon as the port goes offhook. Since Asterisk does not really track the status of FXO ports, what happens is undefined and could change in the future. |
| 16:44.05 | ManxPower | immediate=yes is for Bat Phone types of applications. |
| 16:45.28 | Spirits-Sight | Ok, I reset the phone and now I am going to set the proxy server (the ip address of the system with asterisk right?) and user id and password of extions or not |
| 16:45.36 | Spirits-Sight | making sure understand before I do it |
| 16:45.52 | ManxPower | set the userid and password |
| 16:47.02 | ManxPower | extensions don't normally have passwords. DEVICES have passwords. The fact that you set the device id to be the same as the extension is much like someone named george owning a cat named George. They are NOT the same person, they are just named the same. |
| 16:47.10 | hi365 | when using languageprefix, where does the custom folder belong, per language or at the language level? |
| 16:49.37 | Spirits-Sight | ManxPower: the only area I see to set user id and passwords are under the ext 1 or 2 tabs of the web interface for the phone |
| 16:50.49 | *** join/#asterisk Defraz (n=T0tal@63.228.246.229) |
| 16:51.19 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.66) |
| 16:52.10 | Katty | [TK]D-Fender: heh, i found some old photos of myself. i look like lara croft in one of them /querying |
| 16:52.32 | Katty | [TK]D-Fender: tell me that isn't hilarious. |
| 16:52.47 | *** join/#asterisk aliver (n=aliver@c-71-196-147-164.hsd1.co.comcast.net) |
| 16:55.06 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 16:56.19 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 16:57.48 | [TK]D-Fender | Katty: Not sure I'd say "hillarious" "hilly" yes.... |
| 17:00.29 | aliver | Any known issues with * 1.4 && Cisco IP Phone 7960's ? I'm getting some buzzing in meetme conferences (ztdummy). |
| 17:00.42 | aliver | But it's intermittent. |
| 17:00.50 | [TK]D-Fender | aliver: No. |
| 17:01.24 | aliver | Ok. |
| 17:01.34 | aliver | Well. Damn. |
| 17:02.09 | etm124 | Katty: I could go for those porkchops you posted a few days ago. |
| 17:03.27 | Katty | etm124: (= |
| 17:03.34 | Katty | jbot: block numbers? |
| 17:03.35 | jbot | jumps in front of numbers and prevents any more damage from being done to all the victims in the channel |
| 17:03.43 | Katty | not quite what i was looking for |
| 17:03.50 | Katty | anyone have a wiki page on how to block specific phone numbers? |
| 17:04.19 | *** join/#asterisk sp00k3y (n=sp00k3y@wsip-98-190-136-194.ph.ph.cox.net) |
| 17:04.27 | russellb | you could use the blacklist stuff |
| 17:04.34 | Katty | jbot: blacklist? |
| 17:04.44 | russellb | lookupblacklist used to be an app |
| 17:04.47 | russellb | i can't remember what it is now |
| 17:04.50 | russellb | so many things have changed form, heh |
| 17:06.03 | Spirits-Sight | ManxPower: IT works, |
| 17:06.37 | Spirits-Sight | I tryed to call ext 2 but it says on the phone, shouldn't it ring ext on the phone seeing its setup |
| 17:07.16 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 17:07.53 | aliver | I have two * servers. Server A generates a callfile when a meetme room gets to max. Server B answers in context "meetmebridge" and drops into the same meetme room # via asterisk realtime arch. After this is complete there is no audio bridge between A & B. |
| 17:10.39 | *** join/#asterisk luke-jr (n=luke-jr@2002:18fc:16e6:0:20e:a6ff:fec4:4e5d) |
| 17:11.07 | luke-jr | Does Digium want to know about copyright infringements in Asterisk (going back to at least 1.2 I think)? If so, how? |
| 17:11.37 | jblack | interesting. What do you know? |
| 17:12.07 | luke-jr | the gsm codec is not licensed for redistribution nor modification |
| 17:12.30 | luke-jr | only use |
| 17:12.40 | jblack | That would be a problem. |
| 17:13.12 | luke-jr | I would imagine. |
| 17:13.44 | jblack | Hmmmm. The file has Mark Spencer's name on it, but he says the code is from TOAST. Yeah, there might be a problem here. |
| 17:14.11 | luke-jr | the COPYRIGHT file in the dir contains the original gsm copyright |
| 17:14.37 | jblack | According to the COPYRIGHT file, any use is is permitted. That's perfectly fine. |
| 17:14.42 | jblack | Distribution is "a use" |
| 17:14.44 | luke-jr | use != distribution |
| 17:16.02 | jblack | I'm not part of digium, but I wouldn't worry about it if I were them. |
| 17:16.17 | luke-jr | most licenses differentiate between use and distribution |
| 17:16.30 | luke-jr | and Debian, gNewSense, and Gentoo seem to concur this is a problem |
| 17:16.45 | luke-jr | (it also affects sox) |
| 17:17.25 | jblack | I don't think it's a problem, as the word "any" is included. |
| 17:17.58 | luke-jr | but distribution isn't use, AFAIK |
| 17:18.21 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 17:18.21 | jblack | If they sued, I'm pretty sure it would hold up in court to say "Yeah, we're using it in our distribution of asterisk. Can we go out for coffee now?" |
| 17:19.16 | luke-jr | why? |
| 17:19.46 | luke-jr | there is a large quantity of 'freeware' permitting both use and redistribution, but not allowing modification |
| 17:20.14 | luke-jr | distributing it as source does not affect the legal status itself |
| 17:20.39 | jblack | Because, in this particular case, of the phrase "any use", with no qualifiers. |
| 17:21.29 | jblack | Anything that anyone can define as a use is allowed by that license, with the only restriction that the COPYRIGHT remain intact. |
| 17:22.20 | jblack | If you seriously think it's a problem, then by all means push it. But I wouldn't bother. |
| 17:23.47 | jblack | However, be aware that it's nearly identical to latter gen BSD license, which is pretty well understood to allow everything. |
| 17:27.05 | *** join/#asterisk stimpie (n=stimpie@213-73-211-48.cable.quicknet.nl) |
| 17:27.51 | thedonvaughn | any use is permitted. including the use of distributing it with asterisk. |
| 17:28.12 | tzafrir_laptop | luke-jr, for starters, it is poorly-worded .IT seems that in German it really means that |
| 17:28.33 | lmadsen | luke-jr: you can email jwebster@digium.com about that |
| 17:28.35 | tzafrir_laptop | http://lists.debian.org/debian-legal/2006/02/msg00107.html |
| 17:29.37 | *** join/#asterisk giovani (n=giovani@unaffiliated/giovani) |
| 17:30.05 | *** join/#asterisk korihor (n=korihor@201.210.239.172) |
| 17:30.19 | giovani | would anyone here happen to know where the most recent (albeit old) copy of ARI (Asterisk Recording Interface) might be mirrored/available? |
| 17:30.19 | luke-jr | tzafrir_laptop: the license is not in German |
| 17:30.32 | giovani | littlejohnconsulting's site's been down for a while |
| 17:30.48 | jblack | The computer world is one of the few places that is more pedantic than the law. :) |
| 17:31.29 | tzafrir_laptop | giovani, I figure that the only copy of it that is actively maintained is the one in freepbx |
| 17:31.40 | tzafrir_laptop | jblack, parts of it |
| 17:31.47 | giovani | tzafrir_laptop: I didn't realize they were maintaining it |
| 17:32.01 | tzafrir_laptop | Hopefully they do |
| 17:32.06 | giovani | I thought they just packaged it -- trying to rip it out of their distro seems unclean |
| 17:32.24 | giovani | tzafrir_laptop: any alternatives you'd recommend? |
| 17:32.31 | giovani | for web-based access to voicemail |
| 17:33.47 | *** join/#asterisk hi365_m (n=hi365@213.151.34.153) |
| 17:34.34 | *** join/#asterisk qdk (n=qdk@79.138.231.59.bredband.3.dk) |
| 17:34.54 | luke-jr | Didn't 1.6 add ASR or MRCP or such? |
| 17:36.15 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:41.41 | Spirits-Sight | Why if I have two ext setup would it when call ext 2 (200) would it say that the person at extion 200 is on the phone ... I am using spa 942 phone which is able to have 2 ext |
| 17:42.56 | giovani | Spirits-Sight: because the extension number is 200? |
| 17:43.35 | Spirits-Sight | I am calling using ext 100 |
| 17:43.49 | giovani | heh, that's not what you just said |
| 17:45.45 | Spirits-Sight | oops I was not clear sorry, I am on ext 100 and I am caling ext 200 and its giving me that msg |
| 17:48.03 | giovani | ... you're really not making sense |
| 17:48.15 | Spirits-Sight | however if I go to ext 2 and call ext 1 it appears to work |
| 17:48.20 | giovani | if you call extension 200, and it says extension 200 is busy ... that means asterisk thinks it's busy |
| 17:48.37 | giovani | if it's incorrectly reporting the busy status, turn on verbose logging, and see why |
| 17:52.06 | Spirits-Sight | ok, I change some thing on the phone and now both are working it appears |
| 17:52.32 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 17:54.59 | ManxPower | looks like the copyright file does not exclude distro or copying |
| 17:55.38 | Spirits-Sight | ManxPower: I got it to work by doing what you said |
| 17:56.32 | Spirits-Sight | ManxPower: Is there a provider that you would recommend of the others for allowing unlimitied incoming on toll free number that is not expecive? |
| 17:56.34 | *** join/#asterisk feeds (n=feeds@85-135-231-54.adsl.slovanet.sk) |
| 17:59.52 | ManxPower | Spirits-Sight: No. |
| 18:00.01 | ManxPower | I don't use providers anymore. |
| 18:00.07 | ManxPower | Too unreliable for my requirements. |
| 18:00.28 | *** join/#asterisk Nasra (n=maxshipp@190.166.70.182) |
| 18:00.35 | Spirits-Sight | So if I may ask then how do you have phone numbers and that |
| 18:00.58 | Spirits-Sight | and make calls out and in board |
| 18:01.12 | ManxPower | Spirits-Sight: I use a PRI from the telco. |
| 18:01.15 | giovani | Spirits-Sight: unlimited incoming on a toll-free number? uh, no :) |
| 18:01.46 | giovani | the PRI is connected to a provider |
| 18:02.51 | Spirits-Sight | So what is a good way that does not require hardware to allow unlimited incoming on toll free and cost less then 60 month |
| 18:03.16 | giovani | Spirits-Sight: you will not find "unlimited incoming" on a toll-free DID |
| 18:04.20 | giovani | that would just be a situation ripe for abuse |
| 18:04.27 | *** join/#asterisk kotique (n=picachu@host-static-89-41-72-85.moldtelecom.md) |
| 18:04.31 | kotique | rrr |
| 18:05.15 | Spirits-Sight | right now, with vocalocity I pay 20 for a unlimitied toll free number, and 39.95 for a extion which allows what ever channels (don't know number) |
| 18:05.46 | Spirits-Sight | it comes out to each month 59.95, so is there a way to do this for less |
| 18:06.04 | *** join/#asterisk WimpMan (n=wimpy@213.240.181.251) |
| 18:06.20 | giovani | Spirits-Sight: I hope they get that service abused for making their pricing scheme so stupid |
| 18:06.47 | ManxPower | Spirits-Sight: I have never ever heard of a provider giving "unlimited incoming toll free" I have heard of them doing unlimited OUTGOING toll free. |
| 18:07.07 | giovani | a toll free number is clearly a position for abuse |
| 18:07.14 | ManxPower | outgoing toll free numbers cost the carrier per min charge |
| 18:07.32 | giovani | ManxPower: what does an "outgoing toll free number" mean? |
| 18:07.38 | giovani | you mean, one calling a toll-free number? |
| 18:07.50 | ManxPower | giovani: you calling the american airlines toll free number from your account. |
| 18:08.11 | ManxPower | incoming would be receiving calls on a toll free number that you own. |
| 18:08.20 | giovani | ManxPower: ok ... I wouldn't really compare that to a toll-free DID ... they're completely separate concepts |
| 18:08.53 | ManxPower | "toll-free" is really incorrect. It's really "toll paid for by receiver of call", much like an automated collect call |
| 18:08.54 | giovani | considering the lowest per-minute rates I've seen available to consumers is about 2 cents per minute ... a single channel used 24/7 for a month would amount to like $860 worth of per-minute charges -- any company willing to provide that for $20/mo is going to get their ass kicked |
| 18:09.17 | giovani | (that would be 2 cents per minute on incoming minutes on a toll-free DID) |
| 18:09.19 | Carlos_PHX | What these providers do is simply cancel over-users. |
| 18:09.33 | Carlos_PHX | Everyone selling "unlimited" has language in the contract to limit it. |
| 18:09.51 | giovani | well, not everyone -- but, it's in their best interest, sure |
| 18:10.06 | Carlos_PHX | It sucks because I get customers asking all the time why we don't offer "unlimited." And the answer is because we're not liars. |
| 18:10.11 | giovani | however, a toll-free did with "unlimited incoming" is going to be a much better target for abuse than a regular did, for obvious reasons |
| 18:10.28 | Carlos_PHX | Find me a real "unlimited" provider and I'll hook up and send them thousands of calls. Wanna see if they drop me? |
| 18:10.52 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:11.19 | giovani | Carlos_PHX: I have no doubt they will |
| 18:11.27 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:11.53 | giovani | but, my point is, in a business of overselling -- you have to evaluate the risk of someone going over your (often) unspoken threshhold -- and for toll-free DIDs ... the threat is much larger |
| 18:13.35 | Carlos_PHX | Yes, of course. |
| 18:13.58 | Carlos_PHX | At the carrier level everything is charged for, even PRI minutes aren't totally unlimited, so it's all a game. |
| 18:14.16 | giovani | mmhmm ... but the carriers started the game |
| 18:14.20 | Carlos_PHX | Our underlying charges stretch out to .00001 cents |
| 18:14.25 | Carlos_PHX | Agreed |
| 18:14.25 | giovani | so, to compete, vonage had to continue :) |
| 18:14.36 | giovani | and round and round we go! |
| 18:14.57 | Carlos_PHX | Yeah. The LEC is required to let you do truly unlimited on PSTN, so they can't bump you off. |
| 18:16.37 | orkid | how much does RCF usually cost? |
| 18:16.39 | Spirits-Sight | Carlos_PHX: it seems that Vocalocity does |
| 18:16.47 | orkid | what are the regular prices in NA ? |
| 18:17.07 | Carlos_PHX | Spirits-Sight: Vocalocity does what? |
| 18:17.29 | giovani | Carlos_PHX: off an "unlimited incoming" toll-free DID |
| 18:17.54 | Carlos_PHX | Lots of people offer it. None that I know of will actually delivery it. They will cut you off if you over-use it. |
| 18:18.17 | jblack | I avoid doing recurring business with companies that sell below their cost. |
| 18:18.22 | Carlos_PHX | Let me put it this way, if I found one, I would be wealthy since I'd send them all our trafic. |
| 18:18.52 | *** join/#asterisk xchaos (n=none@CPE0080c828f609-CM0019474d28a8.cpe.net.cable.rogers.com) |
| 18:19.03 | Spirits-Sight | LOL |
| 18:19.20 | kfife | I don't know if they still do it but TelIAX had the STUPIDEST 'unlimited' plan I have ever seen. The plan said 'Unlimited*' in which unlimited meant 2000 (or so) minutes, and by unlimited they meant they wouldn't 'cut you off' they'd just charge you per minute as long as you had prepaid at some per-minute rate |
| 18:20.22 | ManxPower | Spirits-Sight: Go for it. Don't blame us when your provider cuts off your service and holds your number hostage. |
| 18:20.22 | kfife | They called it a 'soft cap' at 2000 (or so) minutes |
| 18:20.22 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 18:20.22 | Qwell | kfife: No providers have unlimited. |
| 18:20.22 | jblack | Teliax got in trouble with me just the other day. They made an unauthorized charge on my card. |
| 18:20.23 | Carlos_PHX | Hopefully we will see use of that word stop, since Verizon has been sued over it on their data plans. |
| 18:20.24 | kfife | I'm going to start a car dealership with the same policy! All cars just $1 with a 'soft cap' at $1. |
| 18:21.05 | kfife | Qwell: correct, but this is an especially flagrant example of walking the line |
| 18:21.37 | kfife | Qwell: especially becasue there is exactly NOTHING unlimited about it. It's not even unlimited by any 'reasonalbe' definition of unlimited. |
| 18:21.39 | jblack | You could always call the department of weights and measures. They're the ones that enforce advertising claims. |
| 18:21.45 | ManxPower | Carlos_PHX: Yeah, but Verizon's "unlimited plan" as "no more than 5GB a month", so basically 1 DVD of data plus some web browsing. |
| 18:21.57 | jblack | huh? |
| 18:21.58 | ManxPower | Yes, my Verizon service was terminated with no appeal. |
| 18:22.20 | kfife | jblack: Actually it's called the FTC |
| 18:22.53 | jblack | kfife: I don't think the bureau of weights and measures is the same company as the FTC. BWM are state agencies, iirc. |
| 18:23.06 | Carlos_PHX | ManxPower: Did they make you pay a termination fee if you had one? |
| 18:23.22 | ManxPower | Carlos_PHX: I did not have to pay a termination feel |
| 18:23.37 | ManxPower | After they lost the class action I got a check for $175 |
| 18:23.42 | kfife | The Federal Trade Commission has exactly the job of making sure that people do not make claims they can not fulfill, and the legal case history creates a fuzzy 'line' across which you are likely to be taken to task. |
| 18:23.43 | jblack | ManxPower: Hmm. I have verizon dsl, and I'm pretty sure I do much, much, much more than 5GB a month. Can you give me moe info? |
| 18:24.06 | Carlos_PHX | jblack: He was talking about Verizon Wireless |
| 18:24.08 | ManxPower | jblack: Sorry, Verizon BROADBAND Access i.e. EVDO over cellular. |
| 18:24.09 | jblack | kfife: And? |
| 18:24.17 | jblack | ManxPower: Ahhhhh. Ok. |
| 18:24.20 | Carlos_PHX | The wireline carriers seem to have a hidden cap in the 150-300gb range. |
| 18:24.34 | ManxPower | Carlos_PHX: I'd be happy with 150GB cap, |
| 18:25.14 | Carlos_PHX | ManxPower: You should start an intentional community where the intention is everyone gets Gb to the house. |
| 18:25.17 | kfife | jblack: If TelIAX were not such a small company (of little consequence), I'd put money that the FTC would take them to task. |
| 18:26.37 | kotique | Hi. How do I enable SIP history ? |
| 18:26.58 | kotique | <PROTECTED> |
| 18:27.16 | kotique | SIP History Recording Enabled (use 'sip show history') |
| 18:27.24 | kotique | but still nothing in show history after several calls |
| 18:27.59 | kfife | BTW, I just looked again out of curiosity. It appears Teliax isn't advertising those rediculous plans anymore. |
| 18:28.31 | Spirits-Sight | Ok, let me reword my question then, is there a Toll-Free that would allow so much calls on the number before changing more money, I don't get that many calls on the Toll-Free number now but that can change and I don't want to be blasted with a high bill, I am trying to save money right now |
| 18:29.20 | florz | Spirits-Sight: how exactly do you expect to save any money by buying an unlimited plan? |
| 18:29.21 | giovani | Spirits-Sight: how many minutes do you expect to do? |
| 18:29.35 | giovani | florz: some unlimited plans will indeed save money, depends on how many minutes you do |
| 18:29.35 | kotique | guys, why are you talking about non-asterisk things on asterisk channel / |
| 18:30.21 | giovani | heh, because it's voip-related |
| 18:30.27 | florz | giovani: rather unlikely, except in a rather small range of usage |
| 18:30.27 | ajohnson | Is it interrupting someone's ability to ask an Asterisk question? |
| 18:30.37 | kfife | Question: I'm running Dahdi for the first time. If my disto gets a new kernel, I have to recompine what: the dahdi-linux.2.x package? |
| 18:30.55 | giovani | florz: nope, I do it all the time with "unlimited incoming DIDs" -- for $8 or so a month |
| 18:31.01 | kfife | ...and NOT the dahdi-tools.x package? |
| 18:31.06 | giovani | break even with per-minute charged DIDs doing 500 minutes a month -- I do far more than that |
| 18:31.40 | Qwell | kfife: correct |
| 18:31.47 | ajohnson | giovani: Get the AUP for your service, I bet someone in there will talk about usage of the service |
| 18:31.57 | florz | giovani: well, you considered all the alternative offerings? Or is it some product with a rather large number of customers? |
| 18:32.10 | tzafrir_laptop | kfife, right |
| 18:32.16 | tzafrir_laptop | dahdi-linux |
| 18:32.27 | giovani | ajohnson: they can talk about it all they want ... they won't be shutting me off, and haven't in years, never heard a peep about it |
| 18:32.38 | giovani | florz: which alternative offerings do you mean? |
| 18:33.04 | florz | giovani: well, whatever is available that you could substitute for your current supplier? |
| 18:33.05 | kfife | and it's just a straight recompile without tweaking the makefile? In other words, the configure script doesn't have to be run to figure out any special iteractions with the new kernel? |
| 18:33.09 | giovani | unlimited services are designed with an average usage in mind, some go over, most go under -- that's how they make profit |
| 18:33.32 | giovani | florz: I don't know what you're talking about? I have accounts with nearly a dozen providers, the pricing is all in the same range, why? |
| 18:35.54 | kotique | well, it's destroying sip history after some time |
| 18:36.07 | Spirits-Sight | Ok, I don't know what the incoming call amount is on the toll free number, I know its not much, most of the calls would be coming on the local number, I do a lot of out going as I use the phone for personal and non-profit stuff, I DON"T want to use the same number for both, I would like to have a ext setup and when using that have my cell number show up as the calling number so that people think I am using my phone (cell) |
| 18:36.22 | kfife | Thanks Qwell, tzafrir_laptop. I remember someone mentioning a somewhat automated way to trigger this recomple. If you know what I'm referring to, can you remind me what it is |
| 18:36.22 | florz | giovani: well, because you break even at 500 minutes at that particular provider's rates, doesn't mean you couldn't get it cheaper somewhere else - plus, yeah, it's somewhat different with markets where you have lots of consumers as customers, where most people simply buy the more expensive unlimited plan because it sounds so nice, and they can pay for you over-using the service that way |
| 18:36.38 | florz | giovani: doesn't apply for toll free inbound, IMO |
| 18:36.43 | Qwell | dkms or something? |
| 18:37.01 | tzafrir_laptop | kfife, dkms attempts to automate that |
| 18:37.06 | giovani | florz: no, it has nothing to do with toll-free inbound, I never said it did -- just saying, that there are circumstances where the "unlimited" plans work out to be cheaper for an individual with a certain usage |
| 18:37.20 | giovani | florz: I haven't seen anyone cheaper -- you're welcome to throw out some names :) |
| 18:37.46 | tzafrir_laptop | kfife, Mandriva tends to support kernel modules through dkms . suse and Debian have their own frameworks |
| 18:37.56 | florz | giovani: well, yeah, that's true, of course - basically, that's heavy users of "consumer-grade products" |
| 18:38.22 | kfife | Qwell, tzafrir_laptop: Thanks. I'm reading up on it now. It sounds like you don't bother with it. Is that true? |
| 18:38.33 | giovani | florz: I wouldn't say, by any stretch that I'm "abusing" the service -- I'm not running a calling card business on it or something -- my usage is probably just higher than the average user |
| 18:39.33 | tzafrir_laptop | kfife, I looked at it a while ago and had some problems, though I don't really recall what |
| 18:39.35 | florz | giovani: well, as long as you are following the contract, you aren't abusing - by "over-use" I just meant that you use it so much that they don't make any money from you |
| 18:39.55 | jameswf | Asterisk needs a cowbell module..... maybe a modification of whisper page... |
| 18:40.12 | giovani | florz: that would require that I know their fees to their carrier -- which I don't |
| 18:40.16 | kfife | Qwell, tzafrir_laptop: Thanks a lot. |
| 18:40.18 | jameswf | press *222 if this conversation needs more coebell |
| 18:40.49 | kfife | jameswf: :) |
| 18:41.32 | florz | giovani: well, if you say that you are using it much more than the break-even point, I'd assume so - I understood that as "more than double that" ... |
| 18:42.01 | giovani | florz: no, the break-even point for a per-minute plan from the same carrier ... break-even for the consumer, not their own costs |
| 18:42.19 | giovani | yeah, I probably do around a thousand minutes inbound on that number |
| 18:42.27 | giovani | well within "consumer" usage |
| 18:43.32 | giovani | I presume the per-minute rates are much higher than they're figuring for the average usage of an unlimited inbound plan because it comes with an unlimited number of channels, so either the customer of the per-minute plan is doing only a handful of minutes per month, or they're doing 10s of thousands, and need the unlimited channels |
| 18:43.33 | florz | giovani: yeah, sure, that's what I understood - but well, just double the break-even point might still be "within range" |
| 18:43.51 | *** join/#asterisk klictel (n=klictel@nat/digium/x-0782b773721f668f) |
| 18:47.58 | *** join/#asterisk jaytee (i=d8cff501@unaffiliated/jaytee) |
| 18:49.26 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:49.29 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
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| 18:52.28 | Katty | http://angela.sleekgeek.org/2008/11/11/blacklisting-numbers/ <- How to Block File. Use at your own risk. |
| 18:52.39 | Katty | ^- Optionall, Qwell |
| 18:52.46 | Katty | s/Optionall/Optionally |
| 18:52.49 | jaytee | hi Katty |
| 18:52.51 | jaytee | cool! |
| 18:52.52 | Katty | hai |
| 18:53.06 | Qwell | blocks file |
| 18:53.54 | [TK]D-Fender | Katty: that'd be "/ignore file" ;) |
| 18:54.44 | jaytee | file, are you in the building? |
| 18:54.59 | file | nope. |
| 18:55.06 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbank.com) |
| 18:55.21 | jaytee | darn |
| 18:56.13 | Katty | pats [TK]D-Fender |
| 18:56.17 | Katty | file: <3 |
| 18:56.26 | *** join/#asterisk riksta (n=rick@92.63.131.41) |
| 18:57.13 | file | tickles Katty |
| 18:57.16 | *** join/#asterisk neurosys (n=neurosys@c-66-229-81-161.hsd1.fl.comcast.net) |
| 18:57.35 | *** join/#asterisk saftsack (n=oliver@g228065032.adsl.alicedsl.de) |
| 18:57.58 | riksta | Hi, when i do an originate via the Manager API I used to call a context in asterisk 1.4 which did a ForkCDR() and then ResetCDR(w) before making the call to the second leg of the originate, this allowed me to have two individual CDRs with properly populated Billable Secs.... when i try this on Asterisk 1.6 I do not get the coorect CDRs, the 2nd CDR always has 0 billable sec..... does anyone have any knowledge of this please? |
| 18:58.22 | kotique | when I run make menuconfig and the save & exit, where are the options written to? |
| 18:58.30 | kotique | *and then |
| 18:59.14 | littlepinkdot | How can I diagnose an outgoing call issue? I have a Digum TDM400P with 4 FXS ports. Incoming calls work fine, outgoing looks like it's dialing but NOTHING happens. |
| 18:59.36 | Qwell | if nothing happens, how does it look like it's dialing? |
| 18:59.59 | littlepinkdot | In asterisk -r, it shows -- Called g0/ww18002252752 -- Zap/1-1 answered SIP/300-08650e50 |
| 19:00.12 | Qwell | then that isn't nothing |
| 19:00.20 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 19:00.22 | littlepinkdot | Hmm =/ Any suggestions? |
| 19:00.31 | denon | fix g0? |
| 19:01.20 | ManxPower | littlepinkdot: What port type? |
| 19:01.24 | littlepinkdot | How? Far as I can see everything is setup correctly. It was working fine last night, nothing since has changed to my knowledge. |
| 19:01.31 | littlepinkdot | ManxPower, fxsks |
| 19:01.39 | littlepinkdot | Analog from PSTN |
| 19:02.04 | ManxPower | littlepinkdot: what happens if you just Dial(Zap/g0/) Do you get a dialtone? If you don' |
| 19:02.22 | ManxPower | t then you either don't have a line plugged into port 1 or the line is bad. |
| 19:02.35 | *** join/#asterisk klictel_ (n=klictel@nat/digium/x-4afa55a0efbe120a) |
| 19:02.39 | littlepinkdot | How exactly do I dial that? =/ And incoming on the same port works fine so the line is working as well as the port. |
| 19:02.52 | ManxPower | littlepinkdot: you modify your dialplan |
| 19:03.35 | ManxPower | instead of whatever Dial line you have that generates "Called g0/ww18002252752 -- Zap/1-1 answered SIP/300-08650e50" change that line to read Dial(Zap/g0/) don't forget the trailing / |
| 19:04.00 | ManxPower | littlepinkdot: what is the CLI output of a working incoming call? Pastebin it. |
| 19:05.27 | *** join/#asterisk bluequijote (n=chatzill@92.4.4.54) |
| 19:06.19 | ManxPower | littlepinkdot: BTW, if your card actually has FXS ports then the ports will blowup when a call comes in. Perhaps you have FXO ports on the card? |
| 19:06.46 | codefreeze-lap | riksta: Hmmm. If the reason for no billsec is no answer time is set, then there are extra options on forkCDR that might help you.... |
| 19:07.09 | luke-jr | FastAGI are not necessarily on localhost, but the Speech API requires local files. How do people usually deal with this? |
| 19:07.19 | riksta | codefreeze-lap: yeah i saw the new forkCDR options, I tried every single one separately ... none worked |
| 19:07.20 | littlepinkdot | ManxPower, http://pastebin.ca/1253536 |
| 19:07.24 | riksta | codefreeze-lap: any other idea? |
| 19:07.41 | rob0 | Pizza! |
| 19:08.03 | littlepinkdot | ManxPower, its worked worked as fxs for the month |
| 19:08.19 | riksta | codefreeze-lap: where do i see the noanswer time? |
| 19:08.23 | riksta | codefreeze-lap: and what is it? |
| 19:09.46 | ManxPower | littlepinkdot: perhaps you forgot that FXO ports use FXS Signaling? |
| 19:10.02 | riksta | codefreeze-lap: i get two CDRs like this: http://pastebin.com/m74022b7b |
| 19:10.17 | riksta | but they both seem the same! |
| 19:10.19 | ManxPower | littlepinkdot: the call came in on ZAP TWO, not ZAP ONE, which is what you are trying to dial out of. |
| 19:10.49 | ManxPower | So, incoming calls to Zap/2 work, but outgoing calls on Zap/1 do not work. You see why this is a totally invalid and useless test. |
| 19:10.52 | littlepinkdot | ManxPower, ah someone must have moved the lines around, thanks for pointing it out. I'll go fix it. |
| 19:11.15 | ManxPower | littlepinkdot: cut their hands off. They should not be messing with your system |
| 19:12.14 | ManxPower | littlepinkdot: Since you are using a GUI I cannot help you further. |
| 19:12.17 | codefreeze-lap | riksta: bad example. All those cdrs have billable seconds set..... |
| 19:12.59 | littlepinkdot | ManxPower, I never touch the GUI |
| 19:13.28 | ManxPower | littlepinkdot: you don't have to touch the GUI. Your incoming call CLI output should be like 5 lines. |
| 19:13.48 | riksta | codefreeze-lap: so they do...sorry my bad.... but what i am saying is that i get one with 0 seconds and one with the right amount set |
| 19:14.39 | codefreeze-lap | riksta: the trick is to see whether the Answer time or the End time isn't set, or both... |
| 19:14.41 | ManxPower | dialedparties.agi is a GUI thing. So is blacklistcheck, user-callerid, |
| 19:14.53 | ManxPower | ~freepbx |
| 19:14.54 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:15.06 | riksta | codefreeze-lap: ok hold, i'll have a look |
| 19:16.17 | *** join/#asterisk koiler (n=jkoyle@5.247.sfcn.org) |
| 19:17.02 | koiler | hi all.. |
| 19:17.27 | koiler | wondering if someone can answer a question regarding iax trunks and call transferring |
| 19:17.59 | codefreeze-lap | koiler: ask and ye shall see |
| 19:18.41 | koiler | I have 3 servers connected via iax trunks. All extensions are SIP extensions. If user A:SystemA dials UserB:SystemB |
| 19:18.42 | *** join/#asterisk kensuke_ (i=c829e4f4@gateway/web/ajax/mibbit.com/x-55ab988b7ddd6ffc) |
| 19:18.50 | koiler | call routes through iax2 trunk fine |
| 19:19.14 | koiler | if userB then transfers the call back to UserC on SystemA - shouldn't SystemB drop completely out of the loop? |
| 19:19.52 | koiler | i.e. UserA -> UserC only via SystemA at that point? |
| 19:20.44 | [TK]D-Fender | koiler: No. |
| 19:21.02 | [TK]D-Fender | koiler: Because ehn you are DIAL-ing around, that leaves * in the middle. There is no "hand-off. |
| 19:21.32 | [TK]D-Fender | koiler: You would have to know that the reason for a call going back to the server its coming from is due to a transfer which it can't. |
| 19:21.45 | [TK]D-Fender | koiler: This leaev that idea pretty much dead in the water |
| 19:21.45 | ManxPower | If everything was SIP or if everything was IAX2 it might work as you want. |
| 19:22.33 | koiler | ok, makes sense. |
| 19:22.33 | [TK]D-Fender | ManxPower: Not even.. just because som user had their phone transfer a call its still only within the dialplan and it is almost 100% certain that it will simply use DIAL tor ead the other server. There won't be any intelligent bridging there.. |
| 19:23.02 | koiler | one more question then - more an architecture/design question |
| 19:23.36 | ManxPower | [TK]D-Fender: depends on the transfer or not. attended transfers create new calls. I think blind transfer does not create a new call. Really the only way to find out is to try it. |
| 19:23.56 | neurosys | If i wanted to tinker with speech recognition and asterisk, where should i start my research? |
| 19:24.07 | koiler | I'd like to route inbound calls (from PRI) to System A to a user in a 2nd office/location. Based on the above, it makes more sense to register a 2nd line for that person on System A |
| 19:24.09 | ManxPower | neurosys: Google is good for that. |
| 19:24.12 | ManxPower | ~mailinglist |
| 19:24.13 | jbot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 19:24.48 | neurosys | ManxPower: :) i know.. but there are several oss projects. any of them work? |
| 19:24.58 | ManxPower | neurosys: As I understand it you have NO chance to make it work unless everything is ulaw or alaw. |
| 19:25.06 | [TK]D-Fender | ManxPower: Any "transfer still just hits the dialplan... the call came in as one channel and would be send to an EXTEN in the dialplan that would "Dial" once more. the fact it goes to the server it came from is irrelevent |
| 19:25.51 | [TK]D-Fender | ManxPower: Might be different if he called the Transfer app, but no way is that likely or viable :) |
| 19:26.25 | *** join/#asterisk mvanbaak (i=vanbaak@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 19:27.14 | ManxPower | [TK]D-Fender: I said for him to try it. |
| 19:27.31 | ManxPower | You might be right, you might be wrong, but he needs to try it regardless. |
| 19:27.33 | [TK]D-Fender | ManxPower: ok/fine/sure/best of luck/yadda/yadda |
| 19:28.18 | koiler | [TK]D-Fender: thanks for the response |
| 19:28.54 | ManxPower | wanders off to try a CentOS install AGAIN. Maybe I've gotten the memory errors fixed. |
| 19:28.55 | koiler | ManXPower: do to the nat'ing involved - using pure SIP will be a pain |
| 19:29.23 | ManxPower | koiler: SIP works JUST FINE with Asterisk and with NAT. You can't reinvite that's about all. |
| 19:29.41 | Katty | ARGGGGHhhhhhhalsdkfalskdjf |
| 19:29.49 | koiler | ManxPower: yeah - I have it working for a few remote users |
| 19:29.50 | Katty | randomly shreds curtains |
| 19:30.09 | koiler | thx all |
| 19:31.35 | *** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net) |
| 19:39.01 | beek | What is Asterisk telling me when it says "Starting simple switch on ..." ? |
| 19:39.02 | *** join/#asterisk _joe (n=joseph@74.51.109.60) |
| 19:40.12 | _joe | hey folks, i've got a few polycom ip 320 phones behind NAT, and they're all connecting to asterisk 1.6 running on a remote server. i have entries like this in sip.conf for each: http://www.pastie.org/312454 |
| 19:40.18 | _joe | i can make outgoing calls just fine |
| 19:40.26 | _joe | but calls between extensions often hit the wrong phone |
| 19:40.40 | _joe | for instance, calls to extension 1 (dave) often go to joe instead |
| 19:40.52 | _joe | i'm guessing it's something to do with the network setup |
| 19:40.53 | ManxPower | beek: that means "hey! something is happening and I'm paying attention to it. Usually an incoming ring or a phone going off hook |
| 19:40.57 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 19:41.00 | _joe | any pointers? |
| 19:41.13 | ManxPower | _joe: Are you using a GUI |
| 19:41.16 | _joe | ManxPower: nope |
| 19:41.35 | ManxPower | _joe: Maybe you have multiple devices registering to the same account? |
| 19:42.00 | _joe | i'm quite certain i don't |
| 19:42.12 | _joe | i have three entries, joe, dave, and adam |
| 19:42.14 | beek | ManxPower: Okay... so that's normal. I have an FXS port connected to my legacy PBX and and FXO port to the PSTN (thus putting * in the middle). If I choose that line from my old PBX and hit a digit I get an immediate fast busy. |
| 19:42.26 | _joe | and each phone has been programmed with its user's name as its extension |
| 19:42.29 | *** join/#asterisk edshupe (n=edshupe@63.164.117.125) |
| 19:42.30 | _joe | in the sip configuration |
| 19:42.42 | beek | ManxPower: It doesn't appear to be getting to the dialplan at all, so I'm trying to figure out why. Thanks! |
| 19:43.51 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 19:45.12 | [TK]D-Fender | _Joe : You have no SECRET therefor no auth... fix this. Also set codecs for your devices, etc |
| 19:45.49 | _joe | [TK]D-Fender: i agree. i was trying to start simple and planned on setting a secret and codecs later. will either of those impact asterisk's confusion of my devices? |
| 19:46.17 | [TK]D-Fender | _Joeyes, esp when transfering calls, etc |
| 19:46.49 | *** part/#asterisk LemensTS (n=matthew@adsl-70-238-180-74.dsl.stlsmo.sbcglobal.net) |
| 19:46.53 | _joe | ok, i'll do that now. thanks for the pointer, [TK]D-Fender :) |
| 19:47.31 | *** join/#asterisk klictel (n=klictel@nat/digium/x-fe6fe65542b9f8ea) |
| 19:51.01 | edshupe | can someone read my post at http://forums.digium.com/viewtopic.php?t=65405 and give me some ideas, please. |
| 19:51.05 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 19:53.00 | [TK]D-Fender | ~freepbx |
| 19:53.00 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:53.01 | [TK]D-Fender | ^^^ |
| 19:53.35 | jblack | It's easy enough with asterisk. I don't know about freepbx. |
| 19:54.39 | edshupe | On all inbound calls, I am trying to ring a list of cell numbers with a prompt to "press 1 to accept" without calling the ones with current calls (avoiding call waiting). |
| 19:55.15 | edshupe | I think that sums it up, I don't need to use FreePBX, I however have no idea about creating contexts, ring groups, etc. |
| 19:55.45 | [TK]D-Fender | edshupe: Then you need to learn to walk before you run. |
| 19:55.47 | [TK]D-Fender | ~book |
| 19:55.47 | jbot | book is, like, Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 19:57.27 | *** join/#asterisk Daejeo (n=chatzill@118.219.208.218) |
| 19:57.40 | *** join/#asterisk awk_r (n=rawk@nat/digium/x-57890c897b2b825b) |
| 19:57.45 | giovani | I have to say ... if the print version of Asterisk: The Future of Telephony 2nd Edition is the same as the PDF version ... there are serious problems |
| 19:57.55 | kensuke_ | Hi, i can set a number of calls per seconds? |
| 19:57.56 | giovani | entire sections duplicated, words missing, horrible misspellings, etc |
| 19:58.42 | Daejeo | Katty: :) meow |
| 20:00.42 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 20:01.33 | Daejeo | is it good idea of playing news on the phone from the RSS/FEED using Asterisk? will people like it |
| 20:01.44 | Daejeo | ? |
| 20:02.02 | *** join/#asterisk c4t3l (i=rcallico@equinox.alluvium.com) |
| 20:02.05 | awk_r | Daejeo yea, i've seen people do it with their email and other random rss feeds |
| 20:02.07 | c4t3l | hello world |
| 20:02.57 | Daejeo | awk_r: cool then i should write a doc |
| 20:03.30 | Katty | Daejeo: hello |
| 20:04.13 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 20:04.23 | *** join/#asterisk RMod_ (n=nicolasj@unaffiliated/rmod) |
| 20:04.43 | Daejeo | awk_r: ATOM supports video |
| 20:05.38 | Daejeo | Katty: :) ah |
| 20:05.58 | Daejeo | Katty: :) you seem to be busy today |
| 20:06.09 | Daejeo | chasing something ? |
| 20:06.17 | Katty | Daejeo: yeah, a bit. |
| 20:06.23 | Katty | Daejeo: my sanity, perhaps. |
| 20:07.05 | *** join/#asterisk [netman] (n=netman@200.Red-88-25-139.staticIP.rima-tde.net) |
| 20:07.23 | awk_r | Katty, so you're running in circles eh? |
| 20:08.28 | Daejeo | awk_r: just imagine TOM and jerry |
| 20:08.51 | awk_r | ah...i miss those toons |
| 20:09.10 | awk_r | Daejeo: (re: ATOM) yea, base64 encoded or as links |
| 20:11.09 | Daejeo | I am going to convert ATOM feed into VoiceXML, will play on the phone. |
| 20:11.37 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:12.45 | Katty | awk_r: i'd prefer to think of my sanity as a straight line leading to nowhere. |
| 20:13.40 | Daejeo | user will get the new entry alert by a phone call. |
| 20:15.46 | awk_r | Daejeo: you need to make a firefox plugin associated with this so a user can press 1 while listening to the feed to go to that website... |
| 20:15.54 | awk_r | that idea was completely theoretical, but would be cool |
| 20:16.22 | awk_r | also under the assumption that the user would be near a computer |
| 20:17.48 | Katty | etm124: http://angela.sleekgeek.org/2008/11/11/pork-chops-yum-yum-crockpot-style/ http://angela.sleekgeek.org/wp-admin/post-new.php?posted=495 |
| 20:18.06 | Daejeo | awk_r: I am using ASR. So. use does not need to press buttons . simply uses can say go to web |
| 20:18.18 | Katty | etm124: erm, http://angela.sleekgeek.org/2008/11/11/homemade-mashed-potatoes-slacker-style/ |
| 20:18.27 | awk_r | Daejeo: yea that works too lol |
| 20:20.55 | *** join/#asterisk skyphyr (n=quassel@92-234-8-140.cable.ubr07.hari.blueyonder.co.uk) |
| 20:21.17 | edshupe | [TK]D-Fender: I came here to learn to walk... |
| 20:22.03 | *** join/#asterisk johann8384 (n=johann83@intra.netlogic.net) |
| 20:22.27 | Katty | gives edshupe a cane. |
| 20:22.27 | ManxPower | edshupe: And I imagine most of us are not going to hold your hand to learn to walk. The Asterisk Book is a good starting point for concepts and design and usage information |
| 20:22.50 | Katty | edshupe: and as an added perk, there are fun diagrams! |
| 20:23.50 | ManxPower | edshupe: What you want to do is fairly complex. Remember Asterisk is not a PBX. Asterisk is a TOOLKIT that lets you BUILD a PBX. |
| 20:24.48 | Katty | Or, alternatively, a migrane. |
| 20:25.58 | Daejeo | awk_r: would you to see the demo in the next couple of days? |
| 20:26.21 | luke-jr | FastAGI are not necessarily on localhost, but the Speech API requires local files. How do people usually deal with this? |
| 20:26.29 | Daejeo | awk_r: which website do you like for listening news ? |
| 20:27.00 | awk_r | Daejeo: sure, and it doesn't matter surprise me |
| 20:28.07 | *** join/#asterisk CrazyTux (n=brandon@user-vcauot0.dsl.mindspring.com) |
| 20:28.32 | Katty | CrazyTux: allo. |
| 20:33.07 | ManxPower | luke-jr: Speech API? |
| 20:33.14 | Deeewayne | waves to Katty |
| 20:33.15 | luke-jr | res_speech |
| 20:33.36 | Katty | Deeewayne: hello thar!! |
| 20:33.39 | Katty | hugs Deeewayne |
| 20:33.51 | Deeewayne | offers cookie to Katty |
| 20:34.22 | Katty | :> |
| 20:34.27 | Katty | did you eated it? :< |
| 20:34.31 | file | luke-jr: you mean the grammars? |
| 20:34.42 | luke-jr | yeah |
| 20:35.34 | Deeewayne | Katty: did you purchase a crazy new animal ? |
| 20:35.53 | Katty | Deeewayne: about 6 weeks ago. |
| 20:35.57 | Katty | Deeewayne: but he's not crazy. |
| 20:36.01 | Deeewayne | they doggy ? |
| 20:36.05 | Katty | nodsnodsnods |
| 20:36.32 | Deeewayne | I got a puppy on the same day as you and returned it one week later |
| 20:37.19 | Katty | why? )= |
| 20:37.26 | *** join/#asterisk devilsoulblack (n=aandaluz@srv.ec-gye.internet.geainternacional.com) |
| 20:38.02 | Deeewayne | was trying to save it but its rightful owner showed up |
| 20:38.10 | Katty | seanbright: how's that centos 5.2 install coming along? |
| 20:38.17 | Katty | Deeewayne: oh :< |
| 20:38.28 | Deeewayne | I was happy because she ate my favorite jeans and my other dogs bed |
| 20:38.36 | Katty | haha |
| 20:38.36 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 20:38.46 | Katty | they can be a handful, that's for sure ;) |
| 20:38.59 | Katty | and iv'e got another year and 6 months to go before riddick simmers down! |
| 20:39.17 | Katty | 1yr8months. |
| 20:39.26 | Katty | Something like that. |
| 20:39.42 | seanbright | Katty: it is not going too well |
| 20:39.49 | Katty | seanbright: )= |
| 20:39.59 | Katty | attempts to bribe centos install with cookies and muffinery. |
| 20:40.17 | ManxPower | seanbright: I just finished my first CentOS install |
| 20:41.02 | Spirits-Sight | What provider would you choice, if you wanted to have "unlimitied" incoming calls (local number) and have a high amount for out going, you want have at less 5 calls able to come in? |
| 20:41.07 | rene- | Hello i am in need of someone who can deliver a t1 router to the bryan tx area in less than 24 hrs |
| 20:41.34 | ManxPower | I'm in need of someone to give me 1 million dollars. |
| 20:41.36 | Spirits-Sight | I have look at ip-com and vitelite and wanted to know if there was any better deals then this |
| 20:41.54 | rene- | please send PV thanks |
| 20:41.56 | [TK]D-Fender | ~itsplist-us |
| 20:41.57 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 20:42.04 | [TK]D-Fender | Spirits-Sight: Go look |
| 20:42.17 | Spirits-Sight | I have been :-) |
| 20:42.44 | Katty | rene-: find someone who will deliver you pizza too |
| 20:43.17 | rene- | hehe that is not very helpful but thanks |
| 20:43.27 | rene- | i am hungry tho |
| 20:43.34 | rene- | but i think it is going to be tacos |
| 20:43.43 | Katty | Fred always liked tacos. Winifred. |
| 20:43.56 | Katty | welcome to #angel-trivia |
| 20:45.31 | rene- | the deal is i have a server in texas that had a digium card with a t1 data and a t1 voice lines happily plugged in |
| 20:45.38 | rene- | till it started behaving bad |
| 20:46.24 | rene- | the t1 data thing worked good but i think a cisco t1 router would be more reliable |
| 20:46.51 | ManxPower | rene-: I sold my only Cisco router *yesterday* |
| 20:46.54 | rene- | damn |
| 20:47.02 | rene- | how much did it go for |
| 20:47.17 | Carlos_PHX | I have plenty of routers, but configuring and getting it out today for overnight...hmmm |
| 20:47.25 | rene- | hey Carlos |
| 20:47.33 | rene- | maybe a couple of days |
| 20:47.36 | rene- | is still ok |
| 20:47.38 | ManxPower | $60, I think |
| 20:47.41 | rene- | no way |
| 20:47.42 | Carlos_PHX | That would be easy. |
| 20:47.43 | rene- | so cheap |
| 20:47.47 | ManxPower | 1720 routers are dirt cheap these days. |
| 20:47.55 | rene- | Carlos_PHX: quote? |
| 20:47.55 | ManxPower | look on ebay |
| 20:47.55 | rene- | heeh |
| 20:48.02 | Carlos_PHX | Yes, we buy them for around $90 all the time. |
| 20:48.08 | Carlos_PHX | Does it need to be rack mount? |
| 20:48.19 | rene- | no |
| 20:48.30 | Carlos_PHX | And is it just a generic T1 data config? |
| 20:48.36 | Carlos_PHX | Or will you do the config? |
| 20:48.39 | ManxPower | You want a Catalyst 5505 cheap? 8-) |
| 20:48.39 | rene- | i used cisco hdlc mode |
| 20:48.45 | Carlos_PHX | The best would be someone local, but finding someone... |
| 20:48.47 | rene- | on the linux router |
| 20:49.01 | rene- | i have somebody at the location who can be your remote hands |
| 20:49.11 | rene- | but we are definitely cisco iliterate |
| 20:49.41 | Carlos_PHX | Ok, with the T1 specs I can do a working config before shipping. |
| 20:49.54 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.42) |
| 20:50.10 | *** join/#asterisk doug (i=doug@breakout.telerama.com) |
| 20:50.12 | seanbright | Katty: the net install is evil. it's not smart enough to know what packages it needs to run the installer. |
| 20:50.14 | Katty | And a Cisco Router in a Pear Tree!!! |
| 20:50.27 | seanbright | either that or the 64bit packaged dependency stuff is screwed more than i expected |
| 20:50.28 | doug | excellent, * channel... |
| 20:50.29 | Katty | seanbright: RUN AWAY!!! RUN AWAY!!!! |
| 20:50.51 | doug | exten => _[a-z-A-Z0-9].,1,Set(todial=${KEYPADHASH(${EXTEN})}) |
| 20:50.55 | doug | that's not quite doing what i would hope |
| 20:51.04 | doug | i.e. let me dial using letters... |
| 20:51.20 | seanbright | doug: remove the - between z and A |
| 20:51.28 | doug | actually, my original was: exten => _[a-z-A-Z0-9].,1,Dial(SIP/$KEYPADHASH(${EXTEN})@ext-sip-account) |
| 20:51.37 | doug | oh yeah, how'd that get in there? |
| 20:51.48 | seanbright | don't ask me, it's your code |
| 20:51.53 | doug | i doubt that's my problem |
| 20:51.54 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 20:52.14 | seanbright | it's not, but that doesn't mean it isn't wrong |
| 20:52.19 | seanbright | :) |
| 20:52.37 | devilsoulblack | how show intensive debug opcion over E1 R2 using openr2 |
| 20:54.46 | doug | i figured someone must have cracked this one, making it easy to dial with letters... |
| 20:55.16 | seanbright | doug: pastebin the relevant portion of your extensions.conf |
| 20:55.18 | seanbright | ~pb |
| 20:55.18 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:55.22 | seanbright | doug: ^^^ |
| 20:58.11 | saftsack | hi, my asterisk connects to another asterisk host via iax. the other host is a dyndns hostname. everytime when the ip of the remote host is changed then the connection breaks and will be never build ab again. |
| 20:58.30 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 20:58.45 | saftsack | is there an option where i can force asterisk to resolve the host everytime the registration is broken? srvlookup seems to be a lightely other thing |
| 20:59.15 | saftsack | "iax2 show peers" shows me the old ip which isnt accessible |
| 23:13.49 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 23:13.49 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.1 (2008/10/09), 1.4.22 (2008/10/02), *-Addons 1.6.0 (2008/10/02), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0+2.0.0 (2008/10/02), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev |
| 23:15.48 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 23:15.48 | *** mode/#asterisk [+o denon] by ChanServ |
| 23:33.33 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 23:42.28 | *** join/#asterisk zecrazytux (n=zecrazyt@boulz.org) |
| 23:42.31 | zecrazytux | hi |
| 23:43.22 | WimpMan | zecrazytux: lo |
| 23:50.35 | *** join/#asterisk Tuxguy (n=homeins6@ip-208-109-154-197.ip.secureserver.net) |
| 23:50.49 | Tuxguy | How often does asterisk look for .call files? |
| 23:50.53 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 23:50.58 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-f4d503413c27c8c0) |
| 23:51.25 | hardwire | I believe it gets notified via the inotify system |
| 23:51.34 | hardwire | so as soon as they are saved |
| 23:51.38 | hardwire | I could be wrong |
| 23:52.09 | *** join/#asterisk luke-jr (n=luke-jr@2002:18fc:16e6:0:20e:a6ff:fec4:4e5d) |
| 23:52.13 | zecrazytux | i've installed asterisk on a server (freebsd, in a jail) and try to create conferences. |
| 23:52.15 | WimpMan | If so that must be optional. |
| 23:52.20 | zecrazytux | I created one |
| 23:52.25 | florz | at least in 1.4 it polls each second, including some race conditions |
| 23:52.27 | zecrazytux | if i join, there's music on hold |
| 23:52.41 | zecrazytux | but when someone else comes, no sound |
| 23:52.46 | hardwire | hmm.. I thought it was more immediate than one second |
| 23:52.57 | zecrazytux | what can be wrong ? |
| 23:53.09 | florz | well, on average it's half a second response time, then :-) |
| 23:56.42 | Katty | hai |
| 23:56.47 | Katty | i haz homemade mashed potatoes. |
| 23:57.47 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 23:58.22 | SwK | Katty, is that like home name concrete? :P |
| 23:59.05 | Katty | SwK: think more like potatoes, butter, milk, and parmesan. |
| 23:59.17 | SwK | heh |
| 23:59.45 | coppice | parmesan is close to concrete |