| 00:02.30 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 00:34.06 | *** join/#asterisk The_REV (~The_REV@c-76-21-1-80.hsd1.ca.comcast.net) |
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| 01:50.37 | KavanS | can I use a tdm410p in a pci-x slot? |
| 01:50.49 | *** join/#asterisk luckyaba (~Lucky@ip72-194-218-169.sb.sd.cox.net) |
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| 05:00.42 | *** join/#asterisk k-man (~k-man@unaffiliated/k-man) |
| 05:01.28 | k-man | with the linksys SPA2102, do you have to do anything special to let it connect to a sip provider over the LAN port rather than the wan port? |
| 05:01.32 | *** join/#asterisk sourcode (~code@ppp-58-8-184-251.revip2.asianet.co.th) |
| 05:07.11 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-qjqaaxfeavizdsis) |
| 05:07.29 | The_REV | why wouldnt you want it connecting thru the WAN port? |
| 05:08.19 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 05:08.37 | WiretapWork | The_REV, if your 'sip provider' is an internal * box |
| 05:09.02 | The_REV | then why wouldnt the wan port still be the uplink port... |
| 05:10.06 | The_REV | dont that the Port names so literally and think about the architecture... |
| 05:15.22 | *** join/#asterisk Aven (~user@95.52.229.140) |
| 05:17.51 | p3nguin | If it's like the 3102, you'll be connecting out the WAN port to your remote ITSP. |
| 05:18.07 | p3nguin | Though you can bridge the LAN and WAN ports together and make it less important. |
| 05:35.51 | k-man | i don't want to use my spa2012 as a router |
| 05:35.59 | k-man | so i hook it to the lan |
| 05:36.06 | k-man | wan port is not connected |
| 05:36.28 | p3nguin | How's that workin' out for you? |
| 05:36.32 | k-man | it was working fine before, but i had to do a factory reset on it. now it seems unable to connect to the sip provider |
| 05:40.32 | p3nguin | Unless you have turned off the dhcpd and set the interfaces to bridged mode, I don't expect it to work. |
| 05:40.46 | k-man | well i turned off dhcp of course |
| 05:40.54 | k-man | i can't see the option to bridge interaces |
| 05:41.58 | k-man | an, in networking service? |
| 05:47.45 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 05:47.57 | k-man | p3nguin: thanks for the pointesr |
| 05:48.19 | k-man | i think i must have been using the wan port before, i just forgot as it was so long since i originaly set it up |
| 05:54.01 | *** join/#asterisk The_REV (~The_REV@70-1-11-183.pools.spcsdns.net) |
| 06:02.00 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:02.01 | schmidts | good morning |
| 06:06.40 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
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| 06:17.32 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:21.32 | *** join/#asterisk Ad-Hoc (~nimbus@62.169.216.185) |
| 06:22.02 | irroot | morning all im guessing its gonna be quiet today our merkan friends have a good holiday |
| 06:29.30 | schmidts | lazy americans :D |
| 06:32.46 | irroot | yeah schmitds whats with that hey getting to work at 3/4 pm :P |
| 06:43.38 | *** join/#asterisk hrolf (~hrolf@202.61.49.9) |
| 06:45.26 | *** join/#asterisk hrolf (~hrolf@202.61.49.9) |
| 06:48.26 | schmidts | but they leave work at 11 pm so i guess its ok then ;) |
| 06:49.17 | irroot | yeah means we have a 7am-11pm day :P |
| 06:52.51 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
| 06:52.53 | schmidts | :D |
| 07:02.20 | *** join/#asterisk kwk (~kleine@carbon.gonicus.de) |
| 07:02.30 | kwk | Hello! |
| 07:02.36 | schmidts | Hi |
| 07:03.26 | kwk | Hi @schmidts: Is there a PHP implementation you know of that is good to query the AMI? I found this one on PEAR but it looks kinda old: http://pear.php.net/package/AsteriskManager |
| 07:05.31 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:06.02 | *** join/#asterisk hetii (~Grzegorz@194.181.154.25) |
| 07:06.07 | hetii | hearbeat |
| 07:07.54 | kwk | I found this library to query the AMI and it requires AJAM: http://www.foriamroot.org/shift8-php-asterisk-manager-library/ What is AJAM? |
| 07:10.11 | *** join/#asterisk hrolf (~hrolf@static-host202-61-49-9.link.net.pk) |
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| 07:16.31 | nightrid3r | ajam = aùi webenabled |
| 07:17.32 | kwk | nightrid3r: found it http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+%28AJAM%29 |
| 07:19.26 | schmidts | hwk sorry i only use it directly with popen and i didnt use a framework for it |
| 07:20.27 | schmidts | just a short question, did you know why i can register to a sip proxy and i also incoming calls are working but for outgoing calls i only get a 407 unauthorized and asterisk didnt try to send the invite again with credentials? |
| 07:21.37 | kwk | @schmidts: maybe you need to configure your proxy to allow outgoing connections |
| 07:24.15 | schmidts | kwk its just form an test asterisk i use, the proxy itselfs works fine, and it also works fine from another asterisk server |
| 07:24.18 | *** join/#asterisk mpe (~mpe@office.ipvision.dk) |
| 07:25.03 | kwk | schmidts: ok then. i'm sorry but i simply have no experience with proxies whatsoever... |
| 07:26.05 | schmidts | :D |
| 07:26.08 | schmidts | thx anyway |
| 07:26.13 | mpe | hi do any of you konw that the sound file "zed" is used for (locatet in the letters directory) ??? |
| 07:26.16 | kwk | you're welcome |
| 07:27.20 | mpe | as I in the progress of translating all the sound files to danish |
| 07:27.26 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 07:27.30 | kwk | mpe: I suppose it's used in conjunction with "festival" a text to speech synthesis tool. |
| 07:30.51 | mpe | ok, one of my colleges, mention that it might be the US / UK version |
| 07:30.57 | mpe | of Z |
| 07:32.43 | *** join/#asterisk nightrid3r (~kvirc@91.176.175.22) |
| 07:35.23 | *** join/#asterisk felimwhiteley (~quassel@109.255.104.145) |
| 07:45.32 | wdoekes2 | mpe: grep is your friend, and it turned up 0 in both 1.8 and 1.4. but indeed, http://en.wiktionary.org/wiki/zed says zed is UK and zee is US |
| 07:46.13 | *** join/#asterisk Assid (~Assid@unaffiliated/assid) |
| 07:46.21 | Assid | heya |
| 07:46.37 | Assid | so im setting up a new hosted asterisk server.. w |
| 07:47.02 | Assid | when i call the number i cant hear anything.. but anything said from that device.. i can hear on the opposite end |
| 07:47.27 | Assid | could anyone shed some light on this |
| 07:48.16 | wdoekes2 | nat issues? |
| 07:48.27 | Assid | yeha i think so |
| 07:48.35 | Assid | was wondering if someone could help me resolve them |
| 07:49.12 | Assid | i have nat=yes , directmedia=no already |
| 07:50.38 | Assid | yeah one way audio |
| 07:51.14 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 07:52.35 | *** join/#asterisk makmak78 (~makmak78@83.145.38.138) |
| 07:54.03 | nightrid3r | did you do port forwarding ? |
| 07:54.20 | Assid | the server isnt part of the nat |
| 07:54.50 | nightrid3r | at youre side ? |
| 07:55.14 | Assid | im behind a nat.. i got 2 devices.. each connecting to different networks |
| 07:55.21 | Assid | i,e, different asterisk servers |
| 07:55.31 | nightrid3r | aha |
| 07:55.46 | nightrid3r | so only 1 will work |
| 07:56.02 | Assid | yeah.. but thats wrong.. beats the purpose of a hosted pbx |
| 07:56.17 | Assid | ive seen hosted pbx's work fine before |
| 07:56.35 | nightrid3r | maybe if you can play around with ports |
| 07:56.48 | nightrid3r | give each device its own port range |
| 07:57.28 | nightrid3r | cos you router has to know what to do with incoeing trafic |
| 07:58.20 | Assid | how do i set it per user |
| 07:58.23 | *** join/#asterisk KindOne (~lol@unaffiliated/kindone) |
| 07:58.28 | *** join/#asterisk djuhl30 (~quassel@121.135.82.142) |
| 07:58.51 | nightrid3r | you set it per lan adress |
| 07:59.10 | djuhl30 | I am not understanding any of the concepts behind VOIP. All I know is I am stationed in South Korea and I want to call the USA cheaply |
| 07:59.30 | Assid | nightrid3r how come there are hosted pbx's that dont need that.. |
| 07:59.44 | nightrid3r | oh |
| 07:59.45 | djuhl30 | I have Ekiga installed and no clue what this SIP protocol is. Can someone point me in the right direction? |
| 07:59.51 | nightrid3r | you could use a stun server |
| 07:59.54 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:00.26 | Assid | can you recommend one i can setup? |
| 08:00.42 | nightrid3r | djuhl30: sip is one of the protocols used to transmit voice data over the internet |
| 08:01.24 | djuhl30 | nightrid3r: ok you register for an account and what can you do with it? Appearantly I am not pluggin it into the softphone correctly to make a call. |
| 08:01.57 | Assid | nightrid3r you ever configured a stun server? |
| 08:02.18 | nightrid3r | Assid: no never did but theres enough stun servers on the net |
| 08:02.18 | *** join/#asterisk hrolf (~hrolf@202.61.49.9) |
| 08:02.52 | nightrid3r | djuhl30: you need to input username, password, sip server and then it should work |
| 08:03.11 | nightrid3r | Assid: stun.ekiga.net |
| 08:03.18 | djuhl30 | Should lol |
| 08:03.30 | djuhl30 | I have tried 3 different accounts lol |
| 08:03.55 | nightrid3r | djuhl30: ekiga is a pain to configure, try x-lite from counterpath |
| 08:04.19 | djuhl30 | But I need to pay to make outbound calls right? |
| 08:04.23 | Assid | i got something from debian packages.. called stun |
| 08:04.29 | djuhl30 | Can I get my own telephone number? |
| 08:04.39 | nightrid3r | djuhl30: sure |
| 08:05.07 | nightrid3r | Assid: yes, if you can put thet on your pbx server it will work to |
| 08:05.30 | djuhl30 | I just need to know the rates |
| 08:05.35 | djuhl30 | I am not living in the USA |
| 08:05.40 | *** join/#asterisk ironm (~ironm@fwj00.e-fon.ch) |
| 08:05.47 | Assid | that goes as the outbound proxy right? |
| 08:05.48 | nightrid3r | djuhl30: store.2600hz.com |
| 08:05.58 | nightrid3r | yes |
| 08:06.38 | *** part/#asterisk kwk (~kleine@carbon.gonicus.de) |
| 08:07.18 | djuhl30 | ekiga couldn't register the account |
| 08:07.25 | djuhl30 | I pulled the info off the email ekiga sent me |
| 08:09.08 | djuhl30 | and are these per minute charges with store.2600hz.com? |
| 08:09.17 | djuhl30 | US dollars? |
| 08:09.25 | nightrid3r | djuhl30: yes $/min |
| 08:09.38 | nightrid3r | or flat rate is also possible |
| 08:10.19 | nightrid3r | but contact them firts cos it might be working only with a pbx |
| 08:10.27 | djuhl30 | I can't tell what this all means in reference to using skype |
| 08:10.34 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
| 08:10.48 | nightrid3r | this has nothing to do with skype |
| 08:11.01 | djuhl30 | I mean price wise |
| 08:11.03 | nightrid3r | its not even compatible with skipe |
| 08:11.07 | djuhl30 | Not the program |
| 08:11.17 | nightrid3r | oh , no idea but i think its allot cheaper |
| 08:11.33 | djuhl30 | All I am seeing in money leaving my pocket when I choose a SIP provider |
| 08:12.06 | djuhl30 | I pay $25 per month to call cellphones around the world and they can't call me back lol |
| 08:12.28 | nightrid3r | yes, there's not much you can do about it, unless you know someone in USA with flatrate who can hook you up to his ip pbx |
| 08:12.29 | djuhl30 | Well they can but they need Skype or make a international phone call |
| 08:14.24 | djuhl30 | But my wife is Korean, and we live in korea. I want her relative to be able to call locally too |
| 08:15.05 | djuhl30 | So I guess I want my cake and to eat it too |
| 08:16.06 | djuhl30 | And I can't even figure out who to pay |
| 08:16.07 | nightrid3r | djuhl30: ok, so you need a DID provider (lets your relatives call in from us and 1 or more local lines and an * box with tdm card |
| 08:16.28 | djuhl30 | nightrid3r: what the hell does all that mean? |
| 08:17.08 | djuhl30 | It sounds all Greek to me. |
| 08:17.17 | nightrid3r | djuhl30: DID = incoming phone number, TDM = card to connect phone lines to a PC |
| 08:17.44 | djuhl30 | Hmm I am learning something every day. |
| 08:17.53 | djuhl30 | I run linux |
| 08:18.18 | nightrid3r | djuhl30: ok thats what * runs on |
| 08:18.46 | djuhl30 | I don't have asterisk installed. |
| 08:19.05 | djuhl30 | I just read a brief article on it in Linux Format Magazine |
| 08:19.13 | nightrid3r | you need to install * on a dedicated pc |
| 08:19.23 | nightrid3r | oh ok |
| 08:19.25 | djuhl30 | I am learning the trial and error way |
| 08:19.34 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 08:19.50 | nightrid3r | then go for the asterisknow iso, that will get you up in like 30 minutes |
| 08:19.56 | djuhl30 | Everything isn't in plain English |
| 08:20.09 | djuhl30 | Nah I have gentoo installed |
| 08:20.29 | djuhl30 | Seems pointless to change this desktop computer to a live distro |
| 08:20.36 | djuhl30 | All my hard work out the window |
| 08:20.46 | nightrid3r | if you have an old pc catching dust you can use that |
| 08:20.50 | djuhl30 | I don't have a TDM card yet |
| 08:20.57 | djuhl30 | nightrid3r: I don't |
| 08:21.09 | djuhl30 | but thanks for the advice |
| 08:21.16 | nightrid3r | whell * will need a dedicated pc |
| 08:21.34 | djuhl30 | Maybe the laptop I have |
| 08:21.41 | nightrid3r | cos * will fight with X11 |
| 08:21.49 | djuhl30 | I just need to get some concepts down |
| 08:22.09 | djuhl30 | nightrid3r: really? * messes with X? |
| 08:22.16 | djuhl30 | wow |
| 08:22.33 | nightrid3r | download the e-book trixbox without tears |
| 08:22.56 | nightrid3r | it will explain allot of the basics |
| 08:23.14 | djuhl30 | Download what? from where? |
| 08:23.29 | djuhl30 | Huh which way did she go George? |
| 08:23.31 | nightrid3r | google for it its a free e-book |
| 08:24.29 | nightrid3r | http://dumbme.mbit.com.au/trixbox2/trixbox2_without_tears.pdf |
| 08:24.40 | djuhl30 | Cool |
| 08:24.44 | djuhl30 | found it |
| 08:25.10 | nightrid3r | its considered the newbie's bible to * |
| 08:25.35 | djuhl30 | cool |
| 08:26.26 | *** join/#asterisk gravin (~gravin@175.139.237.155) |
| 08:26.40 | gravin | hi |
| 08:27.04 | gravin | im using elastix 1.6 and updated freepbx to the latest |
| 08:27.15 | gravin | im having a problem with conference |
| 08:27.27 | gravin | after i entered the pin, it hangsup |
| 08:27.38 | nightrid3r | gravin: #elastix |
| 08:27.38 | gravin | either admin or user is the same |
| 08:27.46 | gravin | oh ok thx |
| 08:28.04 | djuhl30 | I got some reading to do. |
| 08:28.21 | nightrid3r | djuhl30: yeah, absorb the wisdom |
| 08:28.22 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 08:32.25 | djuhl30 | tribox is basically asterisks right? |
| 08:32.31 | djuhl30 | Not assuming anything |
| 08:32.55 | djuhl30 | Centos version of Linux |
| 08:32.58 | djuhl30 | rpm based |
| 08:33.36 | *** join/#asterisk Wiretap (~Wiretap@unaffiliated/wiretap) |
| 08:34.13 | irroot | trixbox is ontop of asterisk != asterisk |
| 08:34.51 | djuhl30 | Same diff thanks |
| 08:35.04 | djuhl30 | was the starting point |
| 08:35.17 | *** join/#asterisk KindTwo (~lol@unaffiliated/kindone) |
| 08:35.28 | *** join/#asterisk k_sze (db4f7a9d@gateway/web/freenode/ip.219.79.122.157) |
| 08:35.47 | djuhl30 | So basically I'd just need the computer and no monitor provided I could get a ssh connection |
| 08:36.12 | djuhl30 | Just a monitor and a keyboard to install |
| 08:37.05 | nightrid3r | right |
| 08:37.33 | djuhl30 | So you can configure it via a web browser? |
| 08:39.04 | nightrid3r | yes |
| 08:40.01 | djuhl30 | I'll see if the wife will let me buy a old computer |
| 08:40.12 | k_sze | You know how, in a POTS, one FXS can serve multiple telephone numbers (not extensions, but actual, registered telephone number)? Can Asterisk handle multiple telephone numbers? Does that depend on the modem? |
| 08:41.17 | djuhl30 | All the options look like a programming language lol |
| 08:41.27 | djuhl30 | I have a hard enough time with bash |
| 08:42.12 | djuhl30 | It'll take me a few days |
| 08:42.22 | djuhl30 | But if I did one page at a time |
| 08:42.29 | djuhl30 | I think i can get it working |
| 08:44.34 | djuhl30 | They have a livecd right? |
| 08:44.38 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 08:44.45 | djuhl30 | or is it an installable version? |
| 08:44.49 | nightrid3r | k_sze: yes it can |
| 08:44.58 | nightrid3r | its all in the dialpan |
| 08:45.12 | nightrid3r | djuhl30: its an install cd |
| 08:45.26 | djuhl30 | ah |
| 08:45.36 | djuhl30 | there goes that idea |
| 08:45.49 | Assid | bah |
| 08:45.55 | Assid | i cant get sipdroid to register |
| 08:45.58 | Assid | :( |
| 08:46.22 | k_sze | nightrid3r: you mean "dialplan", right? |
| 08:46.29 | nightrid3r | Assid: thats strange cos sipdroid works perfect |
| 08:46.56 | nightrid3r | k_sze: yes |
| 08:47.29 | Assid | nightrid3r nat'd device? |
| 08:47.34 | Assid | are you using 1.8 ? |
| 08:48.11 | nightrid3r | Assid: no, i use bluebox |
| 08:48.35 | nightrid3r | last * i used was 1.6 |
| 08:48.52 | Assid | i get unauthorized |
| 08:48.53 | Assid | <--- Transmitting (NAT) to 120.63.x.x:54415 ---> SIP/2.0 401 Unauthorized |
| 08:49.15 | Assid | also get this: <--- Transmitting (no NAT) to 120. |
| 08:49.33 | nightrid3r | your on 3g or wifi? |
| 08:49.38 | Assid | wifi |
| 08:49.45 | Assid | i cant even register with sipdroip |
| 08:49.54 | Assid | basic sip client registers .. but no stun suport |
| 08:49.59 | nightrid3r | so its realy a NAT issue |
| 08:50.07 | Assid | looks like |
| 08:50.13 | Assid | stupid endian firewall |
| 08:50.20 | nightrid3r | it almost always is /( |
| 08:51.58 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 08:52.32 | Assid | how do you suggest i handle this? |
| 08:53.13 | nightrid3r | its very hard to nat multyple devices on the same location |
| 08:53.41 | nightrid3r | look at evry device and check if you can set port ranges |
| 08:53.58 | nightrid3r | if you can for ALL devices you can start configuring your router |
| 08:54.14 | Assid | i need something called source nat? |
| 08:54.18 | nightrid3r | if NOT you can setup a small * box and connect htere |
| 08:54.50 | *** join/#asterisk boazb (~b@bzq-82-80-219-90.red.bezeqint.net) |
| 08:55.13 | nightrid3r | or see if you can use IAX thats supposed to be easy to traverse nat networks |
| 08:55.24 | *** join/#asterisk DanFromUK (DanFromUK@2.27.2.120) |
| 08:55.54 | DanFromUK | Hi, is anyone available to help diagnose an unusual nat issue? |
| 08:56.31 | nightrid3r | hides |
| 08:56.37 | DanFromUK | Everything was fine until we updated from 1.4.35 to 1.4.42, and now our server is responding with 404 Not Found when the remote server sends an OPTIONS ping |
| 08:56.50 | DanFromUK | It might not be a NAT issue. |
| 08:57.39 | Assid | it doesnt authenticate through sipdroid |
| 08:57.42 | Assid | but regular does |
| 08:58.17 | *** join/#asterisk asterisk-learner (~chatzilla@193.227.170.247) |
| 08:58.25 | *** join/#asterisk felimwhiteley_ (~quassel@109.255.104.145) |
| 08:59.22 | *** join/#asterisk Aven (~user@95.52.229.140) |
| 08:59.48 | *** join/#asterisk felimwhiteley (~quassel@109.255.104.145) |
| 09:00.08 | Assid | i dont get it |
| 09:00.10 | Assid | it should work |
| 09:04.11 | Assid | i dont get why the default android app can register.. and sipdroid doesnt |
| 09:04.17 | Assid | does that make sense ? |
| 09:04.42 | djuhl30 | What card did you say I needed? |
| 09:04.46 | Assid | yeah im getting 404 for options as well |
| 09:05.21 | nightrid3r | djuhl30: depends on the type of line you have |
| 09:05.57 | djuhl30 | Does it look like I know what I am doing lol? |
| 09:06.25 | djuhl30 | I scanned that information briefly you told me about. |
| 09:06.34 | Assid | nightrid3r any idea on tthis? |
| 09:06.38 | nightrid3r | djuhl30: for cheap cards atcom or openvox are well known in your area, for high quality go for sangoma or digium cards |
| 09:06.50 | djuhl30 | Now I need to get a plan going to make calls cheaply |
| 09:07.17 | djuhl30 | I need to cut and paste that into a text file so I won't forget. |
| 09:07.27 | nightrid3r | Assid: not realy, i'm thinking, if something comes up i'll let you know |
| 09:09.45 | nightrid3r | djuhl30: start with an * box and a few accounts for testing internal dialing and outside connectivity, once that works you get a card to connect your line(s) |
| 09:10.09 | djuhl30 | Okay |
| 09:10.19 | djuhl30 | I have a second hard drive on this computer |
| 09:10.34 | djuhl30 | I am contomplating whether or not I want to install today |
| 09:11.02 | djuhl30 | If I want to make a phone call I'll just need to boot to asterisk |
| 09:11.20 | djuhl30 | Pain until I can find a cheap computer |
| 09:12.20 | nightrid3r | or if you have enough power run * in a virtualbox |
| 09:13.13 | *** join/#asterisk engrxyz (~fgdfgfdg@212.23.51.7) |
| 09:15.20 | djuhl30 | I never thought about that |
| 09:15.24 | djuhl30 | I should be able too |
| 09:15.32 | djuhl30 | Or XEN thanks for the reminder |
| 09:16.35 | djuhl30 | I'll start by downloading the iso image and making a logical volume for the drive |
| 09:16.48 | djuhl30 | How much hard drive space you think I'll need? |
| 09:17.06 | nightrid3r | 8 gig should be enough |
| 09:17.17 | djuhl30 | Cool |
| 09:19.02 | djuhl30 | 32 bit image should work |
| 09:19.14 | nightrid3r | yes |
| 09:20.07 | djuhl30 | In case I find a cheap computer with a single processor |
| 09:20.14 | djuhl30 | I won't need to download again |
| 09:21.18 | *** join/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 09:27.08 | djuhl30 | I'll need to bridge my ethernet connections with vbox I am guessing |
| 09:27.25 | nightrid3r | yes |
| 09:27.52 | djuhl30 | Cool |
| 09:28.01 | *** join/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 09:28.01 | *** join/#asterisk whirlpbx (~whirlpbx@212.58.196.250) |
| 09:28.05 | djuhl30 | iso is still downloading |
| 09:39.27 | djuhl30 | What does IP PBX mean? |
| 09:43.01 | djuhl30 | What is PSTN? |
| 09:45.18 | nightrid3r | pstn = phone line |
| 09:45.24 | nightrid3r | ippbx = * |
| 09:47.09 | djuhl30 | so basically asterisk routes calls |
| 09:47.31 | djuhl30 | so you can call pretty much anywhere it has a route to. |
| 09:48.07 | nightrid3r | exactly |
| 09:48.31 | djuhl30 | Just use over bloated acronyms |
| 09:48.47 | nightrid3r | yeah |
| 09:48.51 | djuhl30 | I'll need to make a diagram or something to keep what I am trying to do straight. |
| 09:49.05 | djuhl30 | Basically turning a linux box into a VOIP router |
| 09:49.20 | nightrid3r | yes |
| 09:49.20 | djuhl30 | Like using NAT as a router. |
| 09:49.45 | djuhl30 | If I can use IPTABLES... I can do this |
| 09:50.03 | djuhl30 | I just don't know if it'll be worth it lol. |
| 09:50.39 | nightrid3r | you'll know only after you try |
| 09:50.45 | djuhl30 | yep |
| 09:50.54 | djuhl30 | and VBOX will let me make mistakes |
| 09:51.08 | djuhl30 | and not really worry about redoing everything |
| 09:51.16 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 09:51.39 | djuhl30 | Makes sense to do it. I'll always have a internet connection |
| 09:51.51 | djuhl30 | And I never know what country I'll be in |
| 09:51.57 | djuhl30 | And I always want to save money |
| 09:52.04 | *** join/#asterisk coppice (~coppice@m121-202-5-227.smartone-vodafone.com) |
| 09:55.23 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 09:57.21 | nightrid3r | djuhl30: http://www.openvacs.com/sub/sub02_02.php bottom of page examples of tdm cards (korean company) |
| 09:57.52 | djuhl30 | crap I need to have my wife translate |
| 09:58.25 | djuhl30 | min of 128 megs of ram to run huh? |
| 09:58.33 | djuhl30 | I am sure this is outdated a little |
| 09:59.15 | nightrid3r | yes, os needs more |
| 09:59.27 | nightrid3r | at least 256 M |
| 09:59.41 | djuhl30 | cool I'll let vbox have that much |
| 09:59.58 | *** join/#asterisk ihor (~Miranda@194.44.15.90) |
| 09:59.58 | djuhl30 | Think I'll need a swap? |
| 10:00.03 | djuhl30 | I can make one later |
| 10:00.18 | nightrid3r | just make swap now |
| 10:00.23 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 10:00.25 | jacc0 | hi all! |
| 10:00.29 | jacc0 | good morning |
| 10:01.12 | nightrid3r | good morning |
| 10:03.08 | djuhl30 | 19:03 in South Korea |
| 10:03.12 | djuhl30 | you are a little late |
| 10:03.21 | jacc0 | :) |
| 10:03.47 | jacc0 | it just turned monday afternoon over here (netherlands) |
| 10:04.30 | jacc0 | am I late or early? |
| 10:04.36 | jacc0 | :) |
| 10:04.43 | nightrid3r | right on time |
| 10:04.54 | nightrid3r | if you bring food :) |
| 10:06.18 | *** join/#asterisk nightrid3r (~nightrid3@91.176.175.22) |
| 10:09.35 | djuhl30 | Asterisk is intalling in VBOX right now. |
| 10:10.10 | djuhl30 | Says 4 min left. It installs quickly |
| 10:15.47 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 10:22.56 | *** join/#asterisk Poincare (~jefffnode@2001:470:cb24:4::2) |
| 10:23.35 | djuhl30 | Cool |
| 10:23.52 | djuhl30 | I guess see if there are any updates via the docs |
| 10:24.17 | nightrid3r | djuhl30 log in an do yum update |
| 10:24.35 | djuhl30 | Just like redhat huh? |
| 10:24.44 | nightrid3r | yup its centos |
| 10:24.44 | djuhl30 | Is yum configured for all repos? |
| 10:24.51 | nightrid3r | yep |
| 10:26.59 | djuhl30 | upgraded 146 packages |
| 10:27.08 | djuhl30 | I'll have to get used to the scripts |
| 10:28.10 | *** join/#asterisk wonderworld (~ww@port-92-201-215-18.dynamic.qsc.de) |
| 10:28.14 | djuhl30 | sudo installed? do you know? |
| 10:28.29 | djuhl30 | Should I create a local user? |
| 10:28.48 | nightrid3r | you can but you don't need to |
| 10:29.05 | nightrid3r | you only use the cli for system upgrades |
| 10:29.21 | djuhl30 | ah |
| 10:29.30 | djuhl30 | the rest is a web browser |
| 10:29.53 | nightrid3r | oh btw if you upgrade freepbx (the gui) don't use yum after to do upgrades, it will break your freepbx install |
| 10:30.04 | nightrid3r | guess who found out the hard way :) |
| 10:30.41 | djuhl30 | you use a tarball or something? |
| 10:30.53 | djuhl30 | a script to install it right? |
| 10:31.12 | nightrid3r | best is to install from tar |
| 10:31.16 | djuhl30 | Or just use what it has |
| 10:31.25 | nightrid3r | but this is easy to get started and experiment |
| 10:31.26 | djuhl30 | You can get the upgrades from the website? |
| 10:31.36 | nightrid3r | yes |
| 10:31.49 | djuhl30 | Still downloading packages |
| 10:32.04 | djuhl30 | After the upgrade I'll try to figure out what I need first. |
| 10:32.35 | nightrid3r | djuhl30 RTFM |
| 10:32.45 | djuhl30 | lol |
| 10:33.01 | djuhl30 | been while since I heard that. |
| 10:33.15 | djuhl30 | But it doesn't hurt to get some free advice. |
| 10:33.46 | djuhl30 | Don't worry, I am sure I'll have some questions after I read a few things. |
| 10:33.55 | nightrid3r | sure |
| 10:34.27 | nightrid3r | oh but ask them in #asterisknow there they support the GUI |
| 10:34.28 | djuhl30 | I can't see anything in the screen shots they have of freepbx |
| 10:34.32 | nightrid3r | here its pure * |
| 10:34.36 | djuhl30 | so I'll need to look at it |
| 10:34.55 | djuhl30 | oh |
| 10:35.23 | djuhl30 | I guess you can't be lazy.... I suppose I can figure out the command line or so |
| 10:36.03 | nightrid3r | yeah going on the cli allows for some very weird and advanced stuff |
| 10:36.19 | djuhl30 | Cool.. I'll get there |
| 10:36.27 | djuhl30 | One thing at a time |
| 10:37.09 | nightrid3r | and just keep hanging in here, you'l learn allot |
| 10:37.27 | djuhl30 | I need to fight the acroyms |
| 10:37.38 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 10:38.04 | nightrid3r | the book explains some of the, the rest well..., google is your friend |
| 10:38.06 | djuhl30 | ekiga will be the smart phone. I have that |
| 10:38.22 | djuhl30 | Yeah if you use the right words |
| 10:39.00 | nightrid3r | x-lite from counterpath also runs on linux |
| 10:39.01 | djuhl30 | SIP gateway part |
| 10:39.22 | djuhl30 | I'll need one of those next I suppose |
| 10:40.39 | djuhl30 | www.voxalot.com I think I did this. Click the free service |
| 10:41.05 | nightrid3r | there's lots of choice for sip providers |
| 10:41.11 | djuhl30 | ok |
| 10:41.16 | djuhl30 | But I have one now |
| 10:41.30 | djuhl30 | I got the email and the free stuff that goes with it. |
| 10:41.40 | djuhl30 | VOIP Service Providers |
| 10:41.57 | djuhl30 | Those are the people that actually route your call right? |
| 10:42.08 | djuhl30 | And they cost money. |
| 10:42.12 | nightrid3r | freecall is also nice, offer between 30 and 40 free country's |
| 10:42.21 | nightrid3r | yes thats right |
| 10:42.37 | djuhl30 | freecall is SIP or a VSP? |
| 10:42.45 | nightrid3r | sip |
| 10:42.55 | djuhl30 | ok |
| 10:44.20 | djuhl30 | memory alloc (4 bytes) returned NULL. is this a actual error? |
| 10:44.33 | djuhl30 | I can't tell if rpm stopped |
| 10:46.21 | djuhl30 | looks like I need a swap I'll make one with dd |
| 10:47.53 | *** part/#asterisk k_sze (db4f7a9d@gateway/web/freenode/ip.219.79.122.157) |
| 10:48.05 | djuhl30 | or up it on the virtual machine |
| 10:48.31 | nightrid3r | told you you needed swap |
| 10:49.32 | djuhl30 | nightrid3r: I must have missed that advice sorry |
| 10:50.52 | nightrid3r | add swap and try upgrade again |
| 10:52.43 | djuhl30 | dd says memory is exhausted |
| 10:53.00 | djuhl30 | I shut down the machine and raised the memory to 512 |
| 10:53.06 | djuhl30 | dd does the same |
| 10:53.28 | djuhl30 | yum on the other hand make pull through |
| 10:57.42 | *** part/#asterisk Pimmetje (~Pimmetje@s5146b5f9.adsl.wanadoo.nl) |
| 10:59.00 | *** join/#asterisk orn (~orn@rtr1.sh23.sip.is) |
| 11:01.19 | irroot | mmm any objections if i add a option for squelching fax detect in dsp ?? |
| 11:03.57 | djuhl30 | It did |
| 11:04.08 | djuhl30 | So I maybe in the clear as far as swap |
| 11:06.59 | *** part/#asterisk mpe (~mpe@office.ipvision.dk) |
| 11:07.07 | *** join/#asterisk war9407 (war@c-71-62-61-74.hsd1.va.comcast.net) |
| 11:15.02 | nightrid3r | wel you can always fix it later |
| 11:16.40 | djuhl30 | https://www.brainshark.com/digium/hello?&r3f1=38027c2f236763646c7e1d263a20292939727a012634213a63386e791a672927 |
| 11:18.55 | coppice | irroot: what exactly do you mean? |
| 11:19.21 | nightrid3r | djuhl30nice |
| 11:19.25 | irroot | coppice been working on a "faxdetect" app for asterisk |
| 11:19.52 | coppice | and? |
| 11:20.18 | irroot | outside the background detect in channels more a WaitFAX |
| 11:21.02 | irroot | if i process the frame in CNG detect if i have it "squelched" in the detector i can then listen for noise |
| 11:21.33 | irroot | that way i can have it better detect voice/audio + fax |
| 11:22.13 | coppice | define squelched |
| 11:23.01 | irroot | in main/dsp.c if the tone detect had squelch option set any frequency matching will be removed "muted" from the frame |
| 11:23.25 | irroot | so the CNG tone will be replaced with silence once accounted for |
| 11:24.38 | coppice | is that a good thing? |
| 11:25.07 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 11:25.13 | irroot | coppice not always in a fax detect loop its fine but having chunks of audio ripped out not so much :P |
| 11:25.29 | irroot | adding DSP_FAXMODE_DETECT_SQUELCH |
| 11:25.41 | irroot | as a bit value |
| 11:26.03 | irroot | option |
| 11:28.13 | irroot | im trying to detect audio excluding fax tones ... to determine if its a human call |
| 11:28.25 | irroot | works quite well with minimal changes |
| 11:33.30 | *** join/#asterisk fenlander (~fenlander@82.152.81.57) |
| 11:33.30 | *** join/#asterisk binbash_ (~peter@a83-161-240-87.adsl.xs4all.nl) |
| 11:33.58 | coppice | squelch to avoid people hearing has value. squlech to avoid machines hearing may not |
| 11:35.01 | coppice | saying audio == voice is extremely woolly. good in tests. no so good in the real world |
| 11:35.40 | djuhl30 | I guess the username and password is outdated too |
| 11:38.27 | mandla | hello tzafrir, does the Xorcom Astribank, XR0032, have a problem with keeping current configuration in the Asterisk Server, whenever power is cut from the Astribank it loses the dahdi configuration |
| 11:38.28 | djuhl30 | ah google helped |
| 11:38.51 | mandla | irroot, hello. |
| 11:39.05 | irroot | hi there mandla |
| 11:39.38 | mandla | irroot, whats'up? How was your weekend. |
| 11:40.27 | irroot | coppice indeed the app generates a 'f' / 'e' tone appropriately and all goes on besides its in a "wait" loop where audio is droped quitely so no harm no foul |
| 11:40.42 | irroot | mandla was semi quiet .... |
| 11:40.50 | irroot | how things there |
| 11:48.05 | coppice | irroot: do you make serious use of T.38 gateway now? |
| 11:48.37 | irroot | have it at a few sites and its working well |
| 11:48.49 | irroot | using it with ooh323 and hylafax |
| 11:49.01 | irroot | for print -> fax AKA asterisk as a faxserver |
| 11:51.03 | irroot | coppice the problem is for some sites need it for shared lines over wireless |
| 11:51.29 | tzafrir | mandla, hi |
| 11:51.38 | irroot | when a call comes in i need to before calling the endpoint and answering need to process the fax |
| 11:52.00 | mandla | tzafrir, hello. |
| 11:52.13 | djuhl30 | Well I got it installed... I'll talk with the wife about purchasing a VSP |
| 11:54.09 | nightrid3r | ok |
| 11:54.41 | coppice | irroot do you use digium fax or spandsp for the gateway? |
| 11:55.01 | irroot | its the spandsp driver in trunk |
| 11:55.08 | irroot | was commited last week fri |
| 11:55.44 | coppice | does the digium stuff support gateway? |
| 11:56.00 | *** join/#asterisk gravin (~gravin@175.139.237.155) |
| 11:56.08 | irroot | not sure i dont know what the digum stuff does wrt fax :P |
| 11:57.13 | coppice | work badly, in many cases :-) |
| 11:57.51 | irroot | does not comment |
| 11:58.49 | coppice | people tell me "support" == "dunno. do you want your money back?" :-) |
| 12:00.51 | Assid | aaargh |
| 12:00.53 | Assid | stupid nat |
| 12:02.36 | irroot | coppice sounds like apple capetown told someone dont use the product if you dont know how :P |
| 12:03.57 | *** join/#asterisk Ejdesgaard (~hej@212.55.52.2) |
| 12:03.59 | Ejdesgaard | hey |
| 12:04.27 | Ejdesgaard | I just tried to download asteriskNow 32&64 bit from http://www.asterisk.org/downloads but i get a 404 in both cases... |
| 12:04.57 | coppice | irroot: there are some real funky bugs in digium fax |
| 12:06.43 | irroot | coppice seen the complaints and feedback but cant comment |
| 12:08.52 | coppice | irroot: the most interesting thing to me is that if i look at the list of people who appear to use commetrex as their fax engine, their T.38 modes all behave differently |
| 12:13.31 | irroot | mmm intresting i dont use many different versions hylafax/t38modem and linksys so far |
| 12:14.13 | irroot | coppice https://reviewboard.asterisk.org/r/1304/ <- dsp changes im working on mostly janitorial |
| 12:19.40 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 12:24.37 | coppice | irroot: if you want to detect FAXes on outgoing calls, detecting 2100Hz alone is pretty useless |
| 12:31.17 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 12:39.24 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:39.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:45.52 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 12:51.07 | irroot | coppice how so ?? |
| 12:51.39 | irroot | you refering to CED tone ?? |
| 12:51.57 | irroot | the confermation tone on answer |
| 13:23.04 | *** join/#asterisk Ryushin (proxy@cl-412.phx-01.us.sixxs.net) |
| 13:28.34 | mandla | tzafrir, ?? |
| 13:29.04 | tzafrir | The astribank does not keep any configuration |
| 13:29.23 | tzafrir | Firmware and such should be automatically loaded at startup |
| 13:31.51 | mandla | tzafrir, but why is that whenever power is cut from the astribank, like power cable to astribank disconnected, this will require me to reinstall the firmware. |
| 13:33.03 | jacc0 | I have this big problem; asterisk is not responding to any sip packet |
| 13:33.39 | jacc0 | when using tshark on the asterisk machine it shows all the incomming invite and optin packages |
| 13:33.58 | tzafrir | mandla, there's no configuration in the astribank. And yes, it is reset when it loses power |
| 13:34.03 | jacc0 | asterisk doesn't recover from this |
| 13:38.01 | jacc0 | I'm running 1.8.5-rc1 in combination with dahdi and debian squeeze |
| 13:38.46 | jacc0 | It looks like a deadlock : there is nothing in the asterisk log |
| 13:39.03 | jacc0 | asterisk acts like there are no incomming calls |
| 13:39.04 | *** join/#asterisk jeffik (~chatzilla@TOROON63-1176243424.sdsl.bell.ca) |
| 13:39.40 | jacc0 | does anyone have a hit on where to look/ what to check? |
| 13:40.02 | mandla | tzafrir, which means it requires to be always powered up. Which means is an extra cost. |
| 13:40.28 | jacc0 | *hint |
| 13:41.34 | tzafrir | jacc0, what do you see in top? |
| 13:42.14 | tzafrir | also try 'H' (shift h) to switch display threads separately |
| 13:44.50 | jacc0 | everting look normal in top |
| 13:45.02 | jacc0 | no proces is taking up more the 1% |
| 13:45.06 | jacc0 | *then |
| 13:45.59 | jacc0 | I have to wait for the event to happen again to take a look with "H" |
| 13:51.21 | jacc0 | it has happend on 2 different servers (HP DL360 G6) in 2 different networks and even diffferent asterisk version; the other server is running 1.8.4.2 |
| 13:53.00 | jacc0 | we have a tool (monit) to check if asterisk is still accepting packages by sending an options request |
| 13:53.29 | jacc0 | in the logging of the tool (that is running on the asterisk server) it shows: no route to host |
| 13:54.41 | jacc0 | looking at the sourcecode of monit it generates the message when trying to connect to the sip port of asterisk |
| 13:55.32 | jacc0 | it looks like it lost it's binding |
| 13:57.43 | *** join/#asterisk cerberus_za (~coert@196-215-44-30.dynamic.isadsl.co.za) |
| 13:59.24 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:b57b:2736:19d8:f0e0) |
| 14:08.54 | *** part/#asterisk Assid (~Assid@unaffiliated/assid) |
| 14:10.46 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 14:17.58 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:17.58 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:27.01 | *** join/#asterisk lcat (~lcat@187.45.253.242) |
| 14:34.38 | *** join/#asterisk k_sze (~Kira@n219079122157.netvigator.com) |
| 14:34.52 | k_sze | Does asterisk do T.37 (store-and-forward)? |
| 14:35.07 | *** join/#asterisk minaguib (~mina@modemcable098.129-202-24.mc.videotron.ca) |
| 14:48.39 | *** join/#asterisk binbash_ (~peter@a83-161-240-87.adsl.xs4all.nl) |
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| 15:07.55 | *** part/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 15:16.11 | *** join/#asterisk mpe (~mpe@212.45.120.202) |
| 15:19.01 | *** join/#asterisk Assid (~assid@unaffiliated/assid) |
| 15:19.04 | Assid | hey |
| 15:19.31 | Assid | so if someone had to setup a fresh asterisk box now.. what are they better off with 1.8 or 1.4 ? |
| 15:21.41 | leifmadsen | 1.8 |
| 15:21.51 | leifmadsen | 1.8 is the only branch currently with maintenance support behind it |
| 15:23.15 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:26.09 | *** join/#asterisk RickB17 (~RickB17@c-69-242-23-245.hsd1.pa.comcast.net) |
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| 15:35.00 | *** join/#asterisk coppice (~chatzilla@116.92.33.189) |
| 15:38.56 | p3nguin | Unless they need reasonable, working SCCP support... then 1.8 isn't the right choice. |
| 15:39.32 | Assid | hrmm atm .. i was facing a major issue with nat ... |
| 15:39.49 | Assid | sip behind nat.. * on the net |
| 15:40.02 | p3nguin | If that's your only problem, you're not doing that bad. |
| 15:40.39 | mickecarlsson | leifmadsen: I have updated ASTERISK-17288, I think that with the info I have provided it should not be that hard to trace the issue and solve it |
| 15:40.40 | CaptainPants | [ASTERISK-17288] [Status: Open] [Assigned: russell] [regression] Asterisk 1.8x, SIP 484 set HANGUPCAUSE to 111 instead of 28 - https://issues.asterisk.org/jira/browse/ASTERISK-17288 |
| 15:41.05 | Assid | well it refused to work... andmy sipdroid refused to register.. but stock sip wregistered.. but no stun suport |
| 15:42.47 | *** join/#asterisk fulcan (~root@li345-191.members.linode.com) |
| 15:44.18 | fulcan | res_config_sqlite.c:844 cdr_handler: attempt to write a readonly database ???? what DB is this error refering to and how do I fix this? |
| 15:44.38 | *** join/#asterisk irroot (~irroot@197.173.180.33) |
| 15:46.00 | leifmadsen | fulcan: it looks like you're trying to write to a read-only database |
| 15:47.45 | fulcan | leifmadsen Ididn't even know sqlite was even on my system untill this error occured. Where the heck it, who is using it and why? |
| 15:48.11 | leifmadsen | as you can see from the command, it looks like cdr_handler is trying to write, which means you have sqlite configured for your CDRs |
| 15:48.25 | leifmadsen | just noload => res_config_sqlite.so in your modules.conf |
| 15:54.11 | fulcan | leifmadsen nothing in modules.conf about sqlite.so? I know there error says it, but nothing in that file about it. |
| 15:54.27 | leifmadsen | fulcan: of course there isn't because you have autoload=yes |
| 15:54.34 | leifmadsen | that's not what I said to look for |
| 15:54.39 | p3nguin | And that's why you get the error. |
| 15:54.43 | leifmadsen | ^^^ |
| 15:56.43 | fulcan | ^w (aka, search) in nano for 'sql' yields 'no results'. so "<leifmadsen> just noload => res_config_sqlite.so in your modules.conf" isn't going to work.... |
| 15:56.56 | p3nguin | It will work if you do it. |
| 15:56.58 | leifmadsen | why not? |
| 15:57.02 | p3nguin | If you never do it, it won't work. |
| 15:57.05 | leifmadsen | :) |
| 15:57.09 | p3nguin | SO GET TO WORK! |
| 15:57.23 | leifmadsen | noload => res_config_sqlite.so is a literal line you should add to modules.conf |
| 15:57.26 | leifmadsen | then you can restart your asterisk |
| 15:57.39 | leifmadsen | for now you can do: *CLI> module unload res_config_sqlite.so |
| 15:57.46 | leifmadsen | the modules.conf is so it doesn't load again in the future |
| 15:58.09 | fulcan | I added it. |
| 15:58.20 | fulcan | ty |
| 15:58.51 | p3nguin | I bet he didn't add the noload to modules.conf. |
| 15:59.21 | p3nguin | I mean, he IRCs from root. |
| 16:00.45 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 16:00.47 | *** join/#asterisk coppice (~chatzilla@116.92.38.98) |
| 16:00.57 | leifmadsen | p3nguin: aye |
| 16:07.35 | *** join/#asterisk SoulRaven (SoulRaven@79.117.174.14) |
| 16:10.35 | *** join/#asterisk k_sze (~Kira@n219079122157.netvigator.com) |
| 16:10.56 | *** part/#asterisk irroot (~irroot@197.173.180.33) |
| 16:11.06 | *** join/#asterisk irroot (~irroot@197.173.180.33) |
| 16:18.22 | *** join/#asterisk g00gle (~thameema@c-98-248-232-219.hsd1.ca.comcast.net) |
| 16:22.26 | *** join/#asterisk drynish (~drynish@modemcable166.15-200-24.mc.videotron.ca) |
| 16:22.43 | drynish | Is anybody here using Google Voice for making calls |
| 16:22.46 | drynish | 6 |
| 16:22.49 | drynish | ? |
| 16:23.01 | p3nguin | yep |
| 16:23.13 | p3nguin | several people. |
| 16:23.37 | suma | drynish: what is your problem ? |
| 16:23.55 | drynish | humph |
| 16:24.02 | drynish | I don't know how to diagnose it |
| 16:24.17 | p3nguin | Just tell us what's going on. |
| 16:24.41 | drynish | I'm doing a DISA and transferring the extension DIALed, I hear a declick and it come back to the disa |
| 16:25.03 | drynish | this is my extension |
| 16:25.11 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 16:25.49 | drynish | http://pastebin.com/PYPjQ0wy |
| 16:26.22 | drynish | so the phone is connecting to the DISA and right after I'm trying to call as simple as that using google voice |
| 16:26.27 | *** join/#asterisk scalex000 (~chatzilla@186.6.158.93) |
| 16:28.18 | p3nguin | Where is the call coming FROM? |
| 16:28.34 | drynish | I want to establish calls |
| 16:28.38 | p3nguin | Where is the call coming FROM? |
| 16:28.51 | drynish | for now, I'm not trying to receive call |
| 16:29.07 | drynish | one sec, I will come back |
| 16:29.13 | p3nguin | If you aren't trying to call, that's why nothing works. |
| 16:29.14 | scalex000 | hi to all, where can I find a database with all pattern |
| 16:29.43 | p3nguin | Uh... a what, with what? |
| 16:30.17 | p3nguin | Do you mean a list of extension patterns? |
| 16:30.40 | *** join/#asterisk drynish (~drynish@modemcable166.15-200-24.mc.videotron.ca) |
| 16:30.45 | drynish | Hi :) |
| 16:30.47 | drynish | Sorry |
| 16:30.57 | drynish | So what I want is to be able to do some calls through Google Voice |
| 16:30.58 | scalex000 | yes |
| 16:31.26 | drynish | My jabber.conf and gtalk.conf should be ok |
| 16:31.55 | scalex000 | I need to create check the area code the user dial and route to the proper root |
| 16:32.00 | p3nguin | ~dialplan patterns |
| 16:32.00 | infobot | dialplan patterns are explained here: http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 16:32.07 | p3nguin | scalex000: ^^ |
| 16:32.19 | scalex000 | yes I check that |
| 16:32.22 | scalex000 | but i have problem |
| 16:32.27 | p3nguin | drynish: If you aren't trying to call, that's why it doesn't work. |
| 16:32.36 | scalex000 | my country use this area code, 809,829,849 |
| 16:32.44 | p3nguin | yeah? |
| 16:32.54 | drynish | I'm trying to call my home phone number |
| 16:33.05 | scalex000 | but the pattern I wan to use is for united states and canada |
| 16:33.07 | scalex000 | :P |
| 16:33.07 | p3nguin | So you want a pattern that matches only those? |
| 16:33.18 | scalex000 | no |
| 16:33.59 | scalex000 | the problems is when i dial mobile phone |
| 16:34.27 | p3nguin | What are you currently doing that isn't working, and what do you want it to do differently? |
| 16:34.45 | p3nguin | drynish: I'll ask you one more time: Where is the call coming FROM? |
| 16:36.11 | drynish | From my ATA connected through SIP to my asterisk server which is linked to my home phone line. |
| 16:36.30 | drynish | I disconnect the one coming from my phone provider |
| 16:36.48 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 16:37.01 | p3nguin | You're making a call from a phone connected to an ATA... |
| 16:37.13 | drynish | yes |
| 16:37.27 | p3nguin | When you dial a number on the phone, where does the ATA send the call? To Asterisk? To some other place? |
| 16:37.32 | drynish | Asterisk |
| 16:37.44 | drynish | I see it coming through |
| 16:37.52 | p3nguin | And where is Asterisk supposed to send the call? |
| 16:38.16 | drynish | to google voice |
| 16:38.43 | *** part/#asterisk irroot (~irroot@197.173.180.33) |
| 16:38.44 | p3nguin | Okay, you don't use DISA for that. So get rid of DISA. Create a dial plan that sends the call out to Google Voice. |
| 16:38.57 | drynish | Yeah but the ATA is not working correctly |
| 16:39.06 | drynish | The only way it's working is using a DISA |
| 16:39.18 | drynish | As soon as he takes the line |
| 16:39.44 | drynish | he has to send the input to the asterisk server |
| 16:40.13 | *** join/#asterisk catphish (~catphish@gateway.office.atechmedia.net) |
| 16:40.42 | catphish | on an incoming SIP INVITE, is it possible to route calls using the To: line instead of the INVITE line? |
| 16:41.07 | *** join/#asterisk felimwhiteley (~quassel@109.255.104.145) |
| 16:41.28 | drynish | From: Michel Di Croci<sip:100@192.168.3.200>;tag=58074296 |
| 16:41.29 | drynish | To: <sip:4505899569@192.168.3.200>;tag=as579402fc |
| 16:41.33 | drynish | What does that mean? |
| 16:41.38 | drynish | When should it go to gtalk? |
| 16:42.19 | p3nguin | It means if you're going to use DISA to make a call from an ATA to Google Voice, I have no desire to help you with it. That's what it means. |
| 16:43.53 | drynish | :) |
| 16:43.57 | drynish | THanks penguin |
| 16:43.59 | drynish | YOu did a lot |
| 16:44.03 | drynish | I know it seems stupid |
| 16:44.07 | drynish | but I found the issue |
| 16:44.18 | drynish | thanks |
| 16:44.21 | catphish | for some reason, a provider i have just started using is sending "INVITE s@hostname" instead of the dialed number, and putting the dialed number in the To: header |
| 16:44.25 | drynish | :) |
| 16:44.32 | drynish | Don't try to understand my broken dialplan |
| 16:44.50 | drynish | it's my ata which is a zoom 5801... it's a really bad unit |
| 16:45.18 | drynish | who's got really funny config |
| 16:46.34 | p3nguin | You can have the most bad unit made, and that still doesn't permiss you to create ignorant dial plan. |
| 16:46.56 | p3nguin | catphish: Are you using a register statement? |
| 16:47.06 | catphish | p3nguin: yes |
| 16:47.32 | p3nguin | catphish: Do you have /your-phone-number on the right side of the register statement? |
| 16:48.16 | catphish | p3nguin: no, some of my providers route multiple numbers per sip-account to me so it never made sense to add one |
| 16:48.30 | catphish | this provider happens to have one number per sip account |
| 16:48.39 | p3nguin | catphish: Does this particular ITSP send more than one? |
| 16:48.46 | catphish | no |
| 16:48.55 | catphish | i will try adding the number to the register line |
| 16:49.02 | p3nguin | catphish: Add the extension and see if it makes any difference. |
| 16:49.07 | leifmadsen | if you register without requesting an extension number, other end points may default to 's' |
| 16:49.12 | p3nguin | It may not, but it will not hurt to try it. |
| 16:49.20 | leifmadsen | if you want to match on To, use type=user not type=peer or friend (which can match on IP address) |
| 16:49.55 | catphish | thanks :) |
| 16:50.20 | catphish | i am worried that one day i will find a provider that does both |
| 16:50.33 | catphish | in which case maybe type=user is the answer |
| 16:50.35 | p3nguin | Both what? |
| 16:50.51 | acidfoo | leifmadsen, I like your irchost -> documanteur-extraordinaire :) |
| 16:50.55 | catphish | i mean one that routes me multiple numbers *and* sends all the invites with s@ |
| 16:51.05 | p3nguin | type=user isn't going to allow you to send calls out from Asterisk to that device (the ITSP). |
| 16:51.12 | anonymouz666 | acidfoo: is this french? |
| 16:51.15 | acidfoo | yup |
| 16:51.39 | leifmadsen | you'll need type=user and type=peer (peer for outbound, user for inbound) |
| 16:52.06 | anonymouz666 | canada has two parts the english and french one ? |
| 16:52.13 | acidfoo | yup |
| 16:52.32 | *** part/#asterisk Poincare (~jefffnode@2001:470:cb24:4::2) |
| 16:52.48 | p3nguin | leifmadsen: You're saying to use two sip definitions in that scenario? |
| 16:54.35 | catphish | i thought users could send and receive calls? |
| 16:54.41 | catphish | or is there a problem with that? |
| 16:54.52 | p3nguin | type=user only allows calling inbound to asterisk. |
| 16:55.02 | catphish | ok |
| 16:55.16 | p3nguin | If Asterisk tries to send a call through a device that is type=user, it will fail. |
| 16:55.21 | catphish | thanks |
| 16:55.25 | catphish | i may need to do that |
| 16:55.53 | *** join/#asterisk binbash_ (~peter@a83-161-240-87.adsl.xs4all.nl) |
| 16:56.19 | catphish | adding the number to the register line worked by the way for this provider |
| 16:56.32 | catphish | but i might look at using the To: line instead in all cases |
| 16:56.36 | catphish | to be more consistent |
| 16:56.54 | p3nguin | I only have multiple numbers with one ITSP, and they, fortunately, send the calls to the number that was dialed. |
| 16:57.30 | catphish | well i have various upstreams, and i want a configuration that works with all of them really |
| 16:57.36 | catphish | so i can automate things better |
| 16:57.54 | catphish | so will try using a user section for them |
| 16:58.17 | catphish | thanks for the pointers anyway :) |
| 16:58.31 | p3nguin | user will be fine if you only get calls in from them and never want to send out. |
| 16:58.52 | catphish | well i can add a peer section with a different name for the outbound |
| 16:58.57 | catphish | that's no problem |
| 16:59.20 | catphish | assuming asterisk will know which to match |
| 16:59.52 | catphish | how does it know which to match? |
| 17:00.01 | catphish | since peer can also accept incoming calls |
| 17:00.08 | p3nguin | That's my concern. I don't know how to get it to match the user and not the peer, when configuring it that way. |
| 17:00.38 | catphish | didn't peer used to be outgoing only? |
| 17:00.52 | p3nguin | Never since I've used Asterisk, no. |
| 17:01.05 | catphish | i don't really understand the difference between peer and friend tbh |
| 17:01.06 | leifmadsen | p3nguin: yes |
| 17:01.17 | leifmadsen | friend == user + peer |
| 17:01.23 | p3nguin | user and friend match on username. |
| 17:01.32 | p3nguin | peer matches ip/port. |
| 17:01.32 | catphish | but if peer allows both, why have friend? |
| 17:01.45 | leifmadsen | although sometimes using type=friend doesn't cause the same type of matching I've found as using a straight up 'user' |
| 17:02.22 | leifmadsen | it's supposed to match like a user first, then fallback to matching on IP address secondary, but I found that's not the case all the time |
| 17:02.33 | catphish | :| |
| 17:02.39 | leifmadsen | I setup a friend for a customer recently and it doesn't match like it should when using a separate user was the key |
| 17:02.41 | leifmadsen | it's not the hard |
| 17:02.56 | leifmadsen | just use type=user for inbound matching of username and peer for outbound dialing |
| 17:03.17 | leifmadsen | catphish: peer does not allow both |
| 17:03.20 | leifmadsen | friend is both |
| 17:03.30 | leifmadsen | peer only matches on IP address for inbound calls |
| 17:03.38 | leifmadsen | user == inbound |
| 17:03.40 | leifmadsen | peer == outbound |
| 17:03.43 | leifmadsen | friend == user + peer |
| 17:03.58 | leifmadsen | or another way.... user == match on username, peer == match on IP address |
| 17:04.00 | catphish | ok, i wonder what's causing my current peers to match inbound calls then |
| 17:04.05 | p3nguin | Saying "peer == outbound" is very misleading. |
| 17:04.10 | leifmadsen | probably matching IP address |
| 17:04.10 | catphish | i probably have an extremely insecure setup |
| 17:04.20 | leifmadsen | p3nguin: I know, but it's a generalization for someone new to the concepts |
| 17:04.42 | p3nguin | I use peer for all my ITSP configurations, sending and receiving calls through them. |
| 17:04.48 | catphish | so do i |
| 17:04.55 | catphish | which is why i'm confused :) |
| 17:04.56 | leifmadsen | peer is a funny category in that it can be generalized into the "outbound" thought, but can do incoming call matching based on IP |
| 17:05.02 | catphish | and peer is matching my incoming calls |
| 17:05.10 | leifmadsen | p3nguin: yes if you're trying to just match on IP address, that is fine |
| 17:05.12 | catphish | but i'm not sure how it's matching them |
| 17:05.32 | leifmadsen | in my case I had the same IP address for separate numbers that I wanted matched differently, so I had to use type=user in order to allow automatic failover once a number was ported |
| 17:05.33 | p3nguin | I'm satisfied with matching incoming calls by IP address from the ITSP. |
| 17:05.44 | leifmadsen | otherwise the authentication fails because it matches the wrong peer |
| 17:05.49 | catphish | how does it know the IP to match? |
| 17:05.57 | p3nguin | host= |
| 17:05.58 | leifmadsen | catphish: because you specify it |
| 17:06.28 | catphish | ok, in that case mine must be matching on reverse DNS |
| 17:06.33 | p3nguin | In the case of host=dynamic, where the device registers to you, that's how it gets the IP address. |
| 17:06.43 | p3nguin | It matches on IP address, not on host name. |
| 17:06.47 | catphish | my host= is set to a hostname |
| 17:06.58 | p3nguin | That's called forward DNS look ups. |
| 17:07.14 | *** join/#asterisk jeremy_g (jeremy_g@c-19a1e755.022-186-73746f3.cust.bredbandsbolaget.se) |
| 17:07.14 | leifmadsen | reverse is taking IP address and getting the hostname |
| 17:07.21 | catphish | yes |
| 17:07.24 | *** part/#asterisk jeremy_g (jeremy_g@c-19a1e755.022-186-73746f3.cust.bredbandsbolaget.se) |
| 17:07.27 | leifmadsen | asterisk doesn't do that |
| 17:07.30 | catphish | and an incoming call comes from an IP |
| 17:07.31 | *** join/#asterisk jeremy_g (jeremy_g@c-19a1e755.022-186-73746f3.cust.bredbandsbolaget.se) |
| 17:07.45 | catphish | so it must be doing a forward lookup when it starts? |
| 17:07.49 | leifmadsen | yes |
| 17:07.51 | jeremy_g | hi, i would like to create a chat room with voice confernecing support. is it possible using asterisk. |
| 17:07.54 | p3nguin | But Asterisk turns any host name in the host= field into an IP address when it loads the conf. |
| 17:07.57 | leifmadsen | if you look at 'sip show peers' you'll IP addresses |
| 17:08.07 | leifmadsen | jeremy_g: chat room? that's not asterisk |
| 17:08.10 | catphish | which brings back my original problem of preventing my peers from matching my incoming calls |
| 17:08.23 | catphish | since they have a host specified to make outgoing calls |
| 17:08.32 | catphish | to the same host the incoming calls originate from |
| 17:08.40 | leifmadsen | jeremy_g: asterisk trunk has some minor video conferencing support in ConfBridge() now |
| 17:08.51 | jeremy_g | leifmadsen:i have an amazing konference and ConfBridge running on asterisk 1.6 and 1.8 so what's stopping me from a chat room? |
| 17:08.54 | leifmadsen | you could potentially take the messaging branch Russell has been working on |
| 17:09.01 | leifmadsen | jeremy_g: you need messaging support |
| 17:09.34 | leifmadsen | http://svn.asterisk.org/svn/asterisk/team/russell/messaging/ |
| 17:09.36 | jeremy_g | leifmadsen:my * can even become a client to gtalk, 1.8 can even send messages. |
| 17:09.43 | leifmadsen | jeremy_g: please say asterisk, not * |
| 17:09.50 | leifmadsen | using * in a sentence is confusing |
| 17:10.13 | leifmadsen | jeremy_g: sounds like you've got it all figured out then |
| 17:10.13 | catphish | perhaps i can prevent my peer from accepting incoming calls by adding some security to it, since i run right now with insecure=port,invite |
| 17:10.13 | jeremy_g | leifmadsen:oh i was trying to be brief. ok |
| 17:10.13 | leifmadsen | catphish: permit and deny might be what you want |
| 17:11.08 | jeremy_g | leifmadsen:i have got bits and pieces here and there but i am trying to build a paltalk like server but for free. |
| 17:11.19 | leifmadsen | ok |
| 17:11.25 | leifmadsen | I have given you all the information I know |
| 17:11.32 | jeremy_g | here goes russel's branch on my ec2 |
| 17:11.48 | *** join/#asterisk boazb (~b@bzq-82-80-219-90.red.bezeqint.net) |
| 17:12.00 | catphish | leifmadsen: thanks for your help, i will try all of that and see what happens, deny sounds like it may match then reject but i'll play with it |
| 17:12.01 | catphish | :) |
| 17:12.09 | leifmadsen | yes it will |
| 17:12.21 | leifmadsen | if you're matching on IP address, then make sure you only have 1 peer setup for that IP address |
| 17:12.36 | catphish | that's the problem, i want both a peer and a user |
| 17:12.40 | leifmadsen | or else it won't end up working like you think (order does not matter, and you can't control the order IP is matched on) |
| 17:12.45 | leifmadsen | yes so define both a peer and a user |
| 17:12.48 | leifmadsen | you can define both |
| 17:12.55 | leifmadsen | it's not an either-or situation |
| 17:13.06 | catphish | i'm sure i can, but what's to stop the peer matching the incoming call instead of the user |
| 17:13.09 | leifmadsen | asterisk will try to match users first, and if nothing matches, will fall back to IP matching (Peers) |
| 17:13.19 | leifmadsen | ^^^ that |
| 17:13.21 | catphish | oh ok, i didn't know that :) |
| 17:13.22 | catphish | thanks |
| 17:13.25 | *** join/#asterisk manji (~manjiki@ppp-2-84-10-2.home.otenet.gr) |
| 17:16.29 | jeremy_g | leifmadsen:doesthis messaging branch implement SIMPLE? |
| 17:16.42 | leifmadsen | XMPP messaging |
| 17:17.31 | jeremy_g | lovely :) that means i can kick ejabber out |
| 17:18.10 | jeremy_g | offers russellb and leifmadsen an energy drink |
| 17:21.12 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 17:25.48 | scalex000 | lol |
| 17:29.24 | *** part/#asterisk jeremy_g (jeremy_g@c-19a1e755.022-186-73746f3.cust.bredbandsbolaget.se) |
| 17:32.43 | drmessano | ohhh |
| 17:32.48 | drmessano | So the messaging is XMPP |
| 17:32.50 | *** join/#asterisk fenlander (~fenlander@82.152.81.57) |
| 17:32.55 | drmessano | Nice |
| 17:32.57 | *** join/#asterisk jeremy_g (jeremy_g@c-19a1e755.022-186-73746f3.cust.bredbandsbolaget.se) |
| 17:33.17 | jeremy_g | leifmadsen:branch compiled..name the application to use for xmpp messaging, please! |
| 17:33.24 | leifmadsen | jeremy_g: no |
| 17:33.29 | drmessano | lol |
| 17:34.09 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 17:34.33 | jeremy_g | ? |
| 17:34.42 | jeremy_g | drmessano:why are you laughing? |
| 17:35.40 | jeremy_g | i need a reply. |
| 17:37.00 | jeremy_g | leifmadsen:do you mean that the branch is a pre-alpha for xmpp |
| 17:37.32 | leifmadsen | you need to look at the list of functions and configuration files available to you |
| 17:37.33 | jeremy_g | that's bad. i am timing out. |
| 17:37.42 | *** part/#asterisk jeremy_g (jeremy_g@c-19a1e755.022-186-73746f3.cust.bredbandsbolaget.se) |
| 17:38.02 | p3nguin | wtf |
| 17:38.11 | p3nguin | What a weird conversation. |
| 17:38.17 | leifmadsen | you're a weird conversation! |
| 17:38.24 | p3nguin | NO U! |
| 17:38.27 | leifmadsen | ok then |
| 17:40.19 | *** join/#asterisk tyman (~tyler@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 17:41.42 | *** join/#asterisk maxagaz (~maxagaz@89.108.151.226) |
| 17:42.25 | drmessano | I got two things from that convo |
| 17:42.41 | drmessano | (1) I can compile from trunk (2) I am so lost |
| 17:43.10 | drmessano | ~whatnow |
| 17:43.10 | infobot | hmm... now is a good time to tell you that I have 6 gigabytes of data |
| 17:43.39 | maxagaz | how can I check with asterisk which ports of my cards are plugged ? |
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| 17:54.03 | tyman | I have two groups of phones in separate contexts, with only one having international dialing abilities. I want both groups of phones to be able to dial each other. If 'include =>' imports BOTH the specified context's extensions AND the specified context's "include'd" contexts then how do I keep the lower privileged phones from making International calls? |
| 17:55.25 | voxter | In the context that isnt supposed to be able to make calls |
| 17:55.41 | voxter | do an exten => _011.,1,Congestion |
| 17:55.44 | voxter | or something |
| 17:55.48 | tyman | ah... |
| 17:55.56 | tyman | yes... i see |
| 17:55.58 | voxter | any defined exten's will supercede includes, ie, they get matched on first. |
| 17:56.14 | tyman | ok...makes sense. ty |
| 17:56.29 | voxter | the one thing to be aware of, is if you have a second priority in your included context, like _011.,2,Play(somemessage), asterisk will continue parsing until you Hangup() |
| 17:57.58 | tyman | I think u mean "s/context/extension" here right? |
| 17:58.27 | tyman | so do, for instance, a Congestion with next priority Hangup? |
| 18:03.50 | p3nguin | Create one context with a higher priv, which includes international calls and includes the lower priv context. Done. |
| 18:05.57 | tyman | p3nguin: that wont let the lower priv phones call the higher priv phones though right? |
| 18:06.21 | p3nguin | http://pastebin.com/vBaX0Ruu |
| 18:06.56 | p3nguin | Then in your sip.conf, specify either context=phones or context=privphones in each phone entry. |
| 18:07.28 | p3nguin | This is how I do it on my own systems. |
| 18:07.38 | maxagaz | I have added a new PCI-E card (with 8 FXO ports) to my, and now I get this error message in my console : WARNING[6731]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x3c4b6c0 (len 618) to xx.xx.xx.xx:5060 returned -1: Operation not permitted. How can I fix it ? |
| 18:07.42 | p3nguin | Create a hierarchy with the contexts. |
| 18:07.52 | tyman | can 'phones' then call 'privphones'? |
| 18:08.08 | tyman | i can see privphones can call phones |
| 18:08.14 | p3nguin | As long as there are extensions for the phones in 'internal', yes. |
| 18:08.22 | p3nguin | It doesn't say privphones can call phones. |
| 18:08.36 | p3nguin | It says privphones includes everything that phones includes, plus international calling. |
| 18:09.31 | p3nguin | Many people have short extension numbers for all their phones in a context called 'internal'. |
| 18:10.11 | tyman | i see...that's the key...got it...that's clean |
| 18:10.13 | p3nguin | exten => 2011,1,Dial(SIP/tyman-phone,42); for example |
| 18:10.57 | p3nguin | Your 'phones' context contains include statements for every other context that you want all phones to have access to. |
| 18:11.18 | tyman | yep...cool... |
| 18:11.28 | tyman | thx |
| 18:11.30 | p3nguin | Then 'privphones' is a "higher level" that contains all the stuff that 'phones' has, plus anything else you want ONLY them to have. |
| 18:11.48 | p3nguin | You can even make a higher level than privphones if you wanted. |
| 18:12.02 | p3nguin | [priv2phones] |
| 18:12.16 | tyman | the key was having the internal extens in 'internal' and including that |
| 18:12.19 | p3nguin | include => more_important_stuff |
| 18:12.25 | p3nguin | include => privphones |
| 18:12.54 | p3nguin | This example makes priv2phones include everything else, PLUS more_important_stuff that no one else has. |
| 18:13.26 | p3nguin | You just have to start at a common point and build up the hierarchy. |
| 18:15.49 | p3nguin | My three-year-old daughter just asked me if she could have <her name>.com ... |
| 18:15.59 | *** join/#asterisk arielb27 (~chatzilla@pdpc/supporter/active/abatista) |
| 18:16.28 | p3nguin | So I bought it for her. Does that make me an enabler? |
| 18:16.40 | leifmadsen | yes :) |
| 18:16.42 | leifmadsen | could be worse |
| 18:16.56 | leifmadsen | had <his name>.com |
| 18:16.59 | leifmadsen | s/had/has/ |
| 18:17.49 | maxagaz | I've plugged a second FXO card (PCI-E), but it takes the place of the first one, how to change this behaviour ? |
| 18:22.30 | scalex000 | what is mean unable to ring user |
| 18:25.18 | *** part/#asterisk catphish (~catphish@gateway.office.atechmedia.net) |
| 18:29.46 | p3nguin | scalex000: Sounds like something FreePBX would say. |
| 18:30.05 | scalex000 | thank you its okey now, I put a comma |
| 18:31.27 | scalex000 | p3nguin, hey you have more experience on asterisk, I found on free tutorial something like jabber |
| 18:31.39 | scalex000 | p3nguin,who its work |
| 18:31.43 | scalex000 | p3nguin,how its work |
| 18:32.19 | p3nguin | I'm not sure what you're trying to say or ask. |
| 18:39.58 | vsniney | maxagaz: depending on the model there may be jumpers that need to be set to control the order they appear |
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| 18:55.36 | *** join/#asterisk hetii (~hetii@87.99.51.172) |
| 18:56.11 | hetii | Hi |
| 18:57.21 | hetii | i have trouble with t38modem - i`m not able to send fax , on log i see: Media Patch:0x17542700RTP_UDPSession 1, Data port on remote not ready. |
| 18:58.20 | hetii | after few of that message i got BYE sip:... SIP/2.0 message |
| 18:59.04 | hetii | maybe some of you had similar issue and know what can be wrong |
| 18:59.07 | hetii | ? |
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| 19:20.37 | *** join/#asterisk theseb (~theseb@75.108.243.48) |
| 19:20.59 | theseb | does there exist services that I can pay to provide an asterisk conf call service? |
| 19:21.10 | theseb | I especially want one that people can connect to with skype |
| 19:35.28 | leifmadsen | theseb: sure, just use chan_skype (although it may be too late to get into that) |
| 19:35.40 | leifmadsen | oh nope, 22 more days |
| 19:35.52 | leifmadsen | http://store.digium.com/productview.php?product_code=1SFA0001 |
| 19:36.09 | leifmadsen | beyond that the rest of the tools are available in asterisk for free |
| 19:36.26 | coppice | Skype, Death Row edition |
| 19:36.46 | leifmadsen | indeed |
| 19:36.50 | leifmadsen | 2 more years though heh |
| 19:36.50 | tyman | MS is the devil |
| 19:36.57 | leifmadsen | I don't think it was MS |
| 19:37.19 | theseb | leifmadsen: what happens after 22 days? it is not like M$ can alter the source code of your asterisk or skype install |
| 19:37.25 | coppice | they are suddenly doing a lot of updates to skype for Linux, so I guess that's working around to killing it. |
| 19:37.32 | leifmadsen | theseb: read the link I gave you |
| 19:37.49 | leifmadsen | after 22 more days you can't purchase or activate the channel driver |
| 19:37.52 | leifmadsen | it will work for 2 more years |
| 19:38.15 | coppice | leifmadsen: they must have had it in planning, but they also clearly waited for their masters to be in place to make any final decision |
| 19:38.35 | leifmadsen | coppice: my guess is that it was in the works for quite some time -- I think Skype regretted making that deal from the get-go |
| 19:38.43 | theseb | leifmadsen: wait..i don't want to activate or set up anything....who can i *pay* to use their asterisk service? |
| 19:38.44 | leifmadsen | I think it had nothing to do with the MS acquisition |
| 19:38.50 | leifmadsen | theseb: zero idea |
| 19:39.02 | theseb | leifmadsen: k, thanks for the info |
| 19:39.03 | coppice | does "can't activate" mean if you screw up a machine you're screwed? |
| 19:39.11 | leifmadsen | no idea |
| 19:39.17 | leifmadsen | I presume if you've activated it previously you'd be ok |
| 19:39.23 | leifmadsen | that's a digium support question though |
| 19:39.26 | tyman | leifmadsen: who do you think were involved with the decision if not MS? |
| 19:39.28 | coppice | skype has been stumbling around without much direction for ages |
| 19:39.33 | leifmadsen | tyman: the people at skype |
| 19:39.39 | leifmadsen | coppice: agreed |
| 19:39.44 | theseb | Not sure why Skype does NOT want asterisk users |
| 19:39.52 | theseb | Oh noes! more business! |
| 19:40.07 | tyman | theseb: exactly...can't see the motivation for skype...only see motivation for MS |
| 19:40.18 | leifmadsen | theseb: skype has been moving away from the free calling between users and trying to get them to use the paid services. chan_skype does nothing to enable that really |
| 19:40.42 | leifmadsen | tyman: you assume Skype is good and MS is evil when I consider Skype more evil in terms of interoperability |
| 19:40.45 | leifmadsen | it's been like that for years |
| 19:40.51 | coppice | chan_skype is a B2B free calls tool. no revenue in that |
| 19:40.56 | leifmadsen | +1 |
| 19:40.59 | leifmadsen | exactly |
| 19:41.18 | leifmadsen | it enables exactly what they don't want you doing on their network |
| 19:41.28 | theseb | leifmadsen: is there anything other VOIP client app that has the same quality of echo/noise reduction as Skype? |
| 19:41.34 | hetii | did some of you have possibility to check t38 transmision ? |
| 19:41.44 | theseb | leifmadsen: in my experience, nothing compares to skype for users that don't buy decent voip hardware |
| 19:41.44 | leifmadsen | theseb: ilbc? speex? |
| 19:42.16 | hetii | maybe some kind of service that i can register log and send fax that are in 99% sure that works :> |
| 19:42.17 | theseb | leifmadsen: don't forget i'm thinking of joe user who has a cheapo microphone and separate speaker all giving feedback |
| 19:42.28 | theseb | leifmadsen: skype was great for dealing with that |
| 19:42.38 | leifmadsen | theseb: yes I know what you are dealing with |
| 19:42.40 | theseb | leifmadsen: i never saw flash up to that level |
| 19:42.41 | leifmadsen | runs far far away |
| 19:42.43 | tyman | leifmadsen: dont assume skype is not evil too...just couldn't see the motivation for them as it's additional userbase and potentials for revenue...whereas MS has competing products (lync et al) that compete with * |
| 19:42.57 | leifmadsen | theseb: tyman: this is probably the most accurate story of what happened: http://babyis60.wordpress.com/2011/05/25/the-long-slow-death-of-skype-for-asterisk/ |
| 19:43.28 | leifmadsen | tyman: no it's not an additional user base that they want -- asterisk users aren't going to be giving them the revenue stream they want |
| 19:44.00 | leifmadsen | cringes every time someone writes a whole sentence only to make it less readable by using * instead of typing asterisk |
| 19:45.09 | p3nguin | coppice: Are you suggesting that Skype for Linux will be gone pretty soon? |
| 19:45.59 | coppice | p3nguin: do you think otherwise? |
| 19:46.23 | p3nguin | coppice: I have no idea. I use Skype on Linux, and this will impact me significantly if it goes away. |
| 19:46.32 | tyman | leifmadsen: does the digium logo make u cringe? |
| 19:46.47 | leifmadsen | tyman: no just your use of * instead of asterisk |
| 19:46.57 | coppice | leifmadsen: the new skypekit is almost useless. they prohibit you doing pretty much anything you might want to |
| 19:47.10 | leifmadsen | coppice: yep that's what I'd been reading -- not surprising |
| 19:47.22 | leifmadsen | "Hey we have this awesome kit that does all the things you don't want to do!" |
| 19:48.59 | coppice | if they put so many restrictions on skypekit its clear the SIP gateway they are going to introduce will be seriously hobbled as well |
| 19:48.59 | tzanger | leifmadsen: sounds like Windows in general |
| 19:49.40 | leifmadsen | coppice: I gave up on skype quite a while ago |
| 19:49.53 | leifmadsen | and no one I know really uses it anyways |
| 19:50.31 | coppice | I know lots of people who use it frequently - mostly families who are physically spread |
| 19:50.45 | tyman | we use skype group video daily...we've researched alternatives and they all suck...would love to know of a comparable solution |
| 19:50.45 | leifmadsen | and it looks like Asterisk 1.10 is going to have decent video support (i.e. it'll do end to end fine as long as you're not transcoding) |
| 19:51.14 | leifmadsen | tyman: I doubt one exists anywhere near the same price point |
| 19:51.25 | leifmadsen | skype group video == Brady Bunch video right? |
| 19:51.49 | leifmadsen | asterisk 1.10 will have video conferencing with a "follow the talker" approach |
| 19:51.49 | tyman | y...poor mans mcu |
| 19:52.06 | leifmadsen | but that's not the Brady Bunch solution most people want |
| 19:52.10 | tyman | works sufficiently well for business even |
| 19:52.27 | leifmadsen | (because that requires the ability to re-encode the video) |
| 19:52.57 | tyman | much more effective than voice only much of the time |
| 19:52.59 | leifmadsen | so basically some sort of integration with ffmpeg I think |
| 19:53.02 | tzanger | remembers the VSX7000 at his old job.. creepy looking thing |
| 19:53.06 | tyman | what's the ETA on 1.10? |
| 19:53.41 | leifmadsen | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 19:54.13 | leifmadsen | you can pretty much assume a new 1.x release once a year (at AstriCon) |
| 19:54.28 | tzanger | leifmadsen: aren't you in the caledonia area? (for some reason I have it in my head that you are) |
| 19:54.38 | leifmadsen | tzanger: Caledon (much different :)) |
| 19:54.40 | tzanger | ah |
| 19:54.43 | tzanger | yes, it is much different |
| 19:55.00 | tzanger | I was going to say I drove though it this past weekend and it's not where I'd expect you :-) |
| 19:55.01 | leifmadsen | my land is not being occupied :) |
| 19:55.24 | tzanger | leifmadsen: there's a joke in there involving a girlfriend but I'm not going to make it |
| 19:55.29 | leifmadsen | :) |
| 19:56.55 | tyman | what type of hw/codec will be required for 1.10 video? |
| 19:57.08 | tyman | will, say, a bria softphone work? |
| 20:06.02 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 20:06.29 | leifmadsen | tyman: anything that supports h.264 |
| 20:07.03 | tyman | cool |
| 20:07.13 | leifmadsen | asterisk is really just passing it through since it can't decode/encode the video stream |
| 20:08.12 | tyman | but, from your previous comments, sounds like it will switch streams based on audio activity? |
| 20:09.14 | *** part/#asterisk hetii (~hetii@87.99.51.172) |
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| 20:39.38 | vsniney | anyone going to astricon this year? |
| 20:39.38 | *** join/#asterisk phatANhappy (~chatzilla@24.123.223.218) |
| 20:39.52 | p3nguin | Nope, it'll be completely empty. |
| 20:40.14 | vsniney | guess i'll stay home then |
| 20:41.20 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 20:41.33 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:42.19 | phatANhappy | n00b, looking for direction for how to provision Polycom 560 phones for asterisk 1.6, i have the provisioning server working, need guidance on base configs |
| 20:43.04 | *** join/#asterisk ffs (~garland@unaffiliated/ffs) |
| 20:44.25 | phatANhappy | have firmware 3.3 loaded into phone, having trouble with cfg files, |
| 20:53.47 | *** join/#asterisk CyberPony (~CyberPony@cpe-075-181-142-115.carolina.res.rr.com) |
| 20:54.14 | *** join/#asterisk nikola_zg (~nix@193.198.162.13) |
| 20:56.13 | phatANhappy | semi -literate n00b at least, any direction would be helpfull |
| 20:56.42 | nikola_zg | hello |
| 20:57.51 | nikola_zg | would appreciate little help regarding matching a peer |
| 20:58.19 | nikola_zg | i have a client that sends sip traffice to me, but from a range of ip addresses |
| 20:58.45 | nikola_zg | type=peer however makes only one ip possible to define |
| 20:59.11 | nikola_zg | is there a simple solution to this type of situation? |
| 21:00.54 | nikola_zg | i tried defining the peer as type=friend, but then the callerid gets overwritten with his username |
| 21:01.06 | nikola_zg | (on the oubound leg) |
| 21:12.05 | *** part/#asterisk CyberPony (~CyberPony@cpe-075-181-142-115.carolina.res.rr.com) |
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| 22:32.13 | *** join/#asterisk dlublink (~david@206-248-131-51.dsl.teksavvy.com) |
| 22:33.54 | dlublink | 'ast_debug(1, "DTMF digit: %c on %s\n", f->subclass, ast->name);', I want to see this message on the console. If I do "core set debug 100" I will see this message in my console? |
| 22:34.15 | dlublink | or do I have to set some debugging option relating to chan_dahdi ? |
| 22:39.33 | russellb | you also need to enable debug logging on the console in logger.conf |
| 22:39.52 | dlublink | ok |
| 22:40.24 | dlublink | ast_debug() matches the debug keyword in logger.conf ? |
| 22:40.47 | russellb | yes |
| 22:40.55 | russellb | you also need to do "core set debug X" |
| 22:43.09 | Kobaz | happy stuff-blows-up day |
| 22:53.26 | scalex000 | hello guys, I have something, when i put a call on hold from Nortel BCM on Asterisk I can't get back |
| 23:15.56 | WIMPy | scalex000: If you're looking for a way to fix it, that was far too little information. |
| 23:21.22 | scalex000 | hmmm |
| 23:21.34 | scalex000 | which information do you need? |
| 23:23.43 | WIMPy | First of all, how are they connected to each other and at which end are you putting the call on hold. What does (not) happen? |
| 23:23.47 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 23:26.16 | scalex000 | ok |
| 23:27.23 | scalex000 | im using sip, the call was put on hold from nortel and the call was make from nortel to asterisk |
| 23:28.15 | *** join/#asterisk tyman (~tyler@c-71-195-122-71.hsd1.ca.comcast.net) |
| 23:28.34 | WIMPy | I saw something on jira regarding SIP holds. |
| 23:31.13 | tzanger | puts WIMPy in a sleeper hold |
| 23:31.30 | scalex000 | lol |
| 23:32.12 | scalex000 | wimpy, so its problem |
| 23:32.18 | scalex000 | or what? |
| 23:32.21 | WIMPy | doesn't get that. |
| 23:33.04 | scalex000 | wimpy, I mean is it a problem? |
| 23:33.35 | WIMPy | I didn't read that issue. |
| 23:35.44 | scalex000 | wimpy, what can I do? |
| 23:36.19 | WIMPy | Search for it on issues.asterisk.org. |
| 23:40.52 | scalex000 | dmtf not suport inband type when is codec g729 |
| 23:49.04 | scalex000 | wimpy, do you think if I put asterisk debug and verbose I will know what happen |
| 23:49.59 | WIMPy | It you read the sip debug, you should be able to find out. |
| 23:51.23 | scalex000 | wimpy, do you know how to log? |
| 23:51.52 | WIMPy | sip set debug [peer|ip> <.... |
| 23:52.03 | WIMPy | s/>/]/ |
| 23:53.46 | scalex000 | in a file |
| 23:55.47 | WIMPy | Don't know if you can configure it in logger.conf. But you can always use > or |. |