| 00:30.59 | *** join/#asterisk blee (~blee@168.234.205.68.cfl.res.rr.com) |
| 00:31.36 | blee | soooooooo im having an strange issue, and it has to do with my polycom phone trying to send a 200OK on an invite like, 10 times, but asterisk ignores it |
| 00:31.59 | blee | i know its a router issue, but i have looked at the capture and cannot see why/what is wrong |
| 00:33.40 | blee | sip trace shows asterisk def receives the 200ok, but does nothing with it |
| 00:39.19 | Khratos | uum, people, is the Asterisk rpm package installation the one who creates the 'asterisk' user and group? |
| 00:39.47 | Khratos | I've just received this (first time I see something like that): |
| 00:39.47 | Khratos | warning: user asterisk does not exist - using root |
| 00:39.47 | carrar | thats a group a hackers that sneak in and add that entry! |
| 00:39.49 | Khratos | warning: group asterisk does not exist - using root |
| 00:41.21 | Khratos | I'm not sure how to interpret your comment, carrar |
| 00:41.46 | carrar | I'd be offended and retort with some swearing |
| 00:42.45 | Khratos | ... well. |
| 00:43.09 | *** join/#asterisk pdtpatrick (~pdtpatric@mainstwan.farheap.com) |
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| 01:03.06 | carrar | Khratos, compile/install from SOURCE! |
| 01:03.17 | jpsharp | Use the source, Luke. |
| 01:04.07 | carrar | slap 1.8 on a 64bit box and be rollin like a Christmas song in Summer! |
| 01:05.30 | Khratos | carrar: I do that on Slackware, but I *must* use CentOS now as it is a client's requirement (you know they turn funny) |
| 01:06.25 | carrar | Asterisk just works in CentOS |
| 01:06.32 | carrar | should have any issues |
| 01:06.34 | carrar | err |
| 01:06.38 | carrar | shouldn't that is |
| 01:06.55 | carrar | 32 or 64 |
| 01:07.33 | Khratos | 32 |
| 01:07.54 | ChannelZ | 110 or 220 |
| 01:08.03 | carrar | 208! |
| 01:08.54 | Khratos | Installation doesn't stop, but Is the first time I see those error lines. |
| 01:09.16 | carrar | kick start that bad boy! |
| 01:10.18 | carrar | I'd assume most people probably run Asterisk as root |
| 01:11.55 | *** join/#asterisk chalcedony (~llhull@unaffiliated/chalcedony) |
| 01:13.18 | carrar | blee |
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| 01:13.27 | carrar | Why do you think it's a router isue |
| 01:13.29 | blee | heyo |
| 01:13.46 | carrar | router just passes packets |
| 01:13.57 | blee | well this issue seems to have sprouted after a replacement router |
| 01:14.43 | carrar | Perhaps you are doing NAT or some firewalling or protocol mangelign on the router |
| 01:15.03 | carrar | which at that point it's more then JUST a router |
| 01:15.05 | blee | Well i realized theres a "SIP helper" that i never tried turning off |
| 01:15.14 | carrar | turn that off |
| 01:15.33 | blee | I guess ive been so focused at looking @ the captures |
| 01:15.36 | blee | which looks perfect |
| 01:15.40 | blee | that i didnt think it was necessary |
| 01:15.43 | carrar | which is good fun |
| 01:16.10 | carrar | incombination with RFC 3261 |
| 01:16.40 | blee | haha yeahhh :-( |
| 01:17.14 | blee | ill turn it off, we'll see that how goes |
| 01:19.38 | *** join/#asterisk fireman_biff (~biff@65.48.132.153) |
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| 01:36.11 | bsdbandit | hello im a newbie looking to learn voip by setting up home voip solution in my house have googling around ive seen many mentions of asterisk which im going to install on openbsd. my question is will i need anything else other than asterisk to allow my self to make phone calls via the internet ? |
| 01:36.44 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 01:39.29 | WiretapWork | bsdbandit, a phone, and an account with an ITSP |
| 01:50.46 | bsdbandit | do you know of any good ITSP worth looking into ? |
| 01:58.33 | WiretapWork | bsdbandit, that is entirely place on the planet dependant |
| 02:09.55 | bsdbandit | oh ok cool |
| 02:10.07 | bsdbandit | thanks for the info |
| 02:12.06 | *** join/#asterisk luckman212 (~irc@2001:470:1f07:1225:90fd:d3bc:f1ac:ec98) |
| 02:13.30 | luckman212 | arg I am tearing out hair |
| 02:13.47 | luckman212 | i do not have enough hair left to tear more |
| 02:21.50 | ChannelZ | get some of that spray-on stuff |
| 02:22.49 | bsdbandit | lol |
| 02:22.50 | bsdbandit | cool |
| 02:26.30 | ChannelZ | I guess there was no followup question there afterall? |
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| 02:47.30 | luckman212 | arg |
| 02:49.21 | luckman212 | so i've got a desk phone, and 2 softphones |
| 02:49.44 | luckman212 | x703, x704, x705 |
| 02:49.55 | luckman212 | i only want to have 1 vm mailbox for these 3 devices |
| 02:50.59 | luckman212 | i thought my answer was to set "mailbox=703@default" for the 704,705 extens |
| 02:51.08 | luckman212 | but this does not work |
| 02:51.42 | luckman212 | whats the difference between "mailbox=" and "vmexten=" ? |
| 02:57.07 | *** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
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| 03:01.01 | fireman_biff | luckman212: http://www.voip-info.org/wiki/view/Asterisk+phone+snom - "Asterisk sends notifications on voicemail messages (if you configured the mailbox option in sip.conf. The messages are sent by default from "asterisk@yourlocalserver.com", which can be modified using vmexten= in sip.conf. When pressing the MWI or Vmail soft button on the SNOM phones the phone calls this extension to connect to the voicemail application" |
| 03:01.10 | fireman_biff | also check http://www.voip-info.org/wiki/view/Asterisk+sip+mailbox |
| 03:01.38 | fireman_biff | thats just for the last question, i'm not sure what to do about the problem itself |
| 03:01.55 | ChannelZ | what do you mean by "does not work" - are you talking specifically about MWI? |
| 03:02.28 | luckman212 | nope... i'm saying I call x704, leave a message and it does not get left in the x703 mailbox |
| 03:03.09 | luckman212 | even though I have set mailbox=703 for that extension |
| 03:03.14 | ChannelZ | well what mailbox it goes in depends on what you call VoiceMail() with |
| 03:03.59 | luckman212 | i'm using freepbx 2.8 |
| 03:04.10 | ChannelZ | well there's your first problem |
| 03:04.17 | ChannelZ | in which case I dunno how to help you then. |
| 03:04.56 | ChannelZ | but mailbox in sip.conf controls what mailbox it queries for MWI indications |
| 03:06.10 | luckman212 | ah i see. maybe thats why. i just assumed that it also set which mbox to use for recordings |
| 03:07.54 | ChannelZ | no. I'm sure FreePBX always calls VoiceMail() with the extension that was dialed. There might be a way to hotwire it properly but you'd have to ask #freepbx |
| 03:08.44 | luckman212 | yes i did find a howto where they describe symlink'ing the directories so they are all basically pointing to the same place |
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| 03:09.45 | ChannelZ | That sounds interesting, I wonder what happens when multiple people are fooling around in the mailbox at the same time. |
| 03:11.08 | luckman212 | i would imaging a MESS |
| 03:11.22 | luckman212 | *imagine |
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| 04:15.19 | luckman212 | ok, the symlinks worked. |
| 04:15.33 | luckman212 | 703,704,705 all sharing the same vm mailbox now :) |
| 04:15.43 | luckman212 | wasnt too messy after all, just about 3 commands |
| 04:21.19 | *** part/#asterisk acidjazz (acidjazz@notchill.com) |
| 04:45.52 | WIMPy | Synlinked mailboxes will probably make the MWIs fail. |
| 04:46.46 | golikwid|mac | couldnt you just point the vm notification to the additional extensions? |
| 04:47.32 | golikwid|mac | in the sip.conf... |
| 04:47.53 | golikwid|mac | something like mailbox=desired mailbox@default |
| 04:48.09 | golikwid|mac | @device.. |
| 04:54.18 | ChannelZ | How does one say the name PrzemysÅaw ? I've never even seen the l with the tick through it before |
| 04:56.10 | fireman_biff | ChannelZ: http://www.forvo.com/word/przemys%C5%82aw/ |
| 04:58.48 | ChannelZ | Interesting. Thanks fireman |
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| 05:18.44 | *** join/#asterisk emsLinux (~emsLinux@190.251.105.192) |
| 05:20.14 | emsLinux | good night, i need some help configuring ODBC voicemail storage, does anyone knows a good tutorial to make it happen, i get the same error all the time "Failed to obtain database object for 'asterisk'" |
| 05:22.01 | kaldemar | emsLinux: http://ofps.oreilly.com/titles/9780596517342/asterisk-DB.html#I_section12_tt1578 |
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| 05:31.37 | GreatSUN | re |
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| 05:54.25 | emsLinux | =( keep getting this message (Failed to obtain database object for 'asterisk'!) from mysql everytime i try to leave a voicemail message, anyone knows a solutions? |
| 05:57.26 | *** join/#asterisk tekzilla (~jon@d230039.adsl.hansenet.de) |
| 05:57.47 | ChannelZ | Set fire to the whole damn place! |
| 05:58.13 | kaldemar | emsLinux: what does your res_odbc.conf look like? pastebin it. |
| 05:58.42 | tekzilla | can i export the sqlite cdr db to csv or easily convert it ? |
| 05:58.51 | emsLinux | Pastebin.com is under heavy load right now, sorry... |
| 05:58.51 | emsLinux | But we are working on it! =S |
| 05:59.01 | kaldemar | ~pb |
| 05:59.01 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 05:59.08 | kaldemar | there are others too. |
| 05:59.23 | emsLinux | i c |
| 06:02.17 | kaldemar | and what is the full warning you're getting? |
| 06:02.47 | emsLinux | http://pastebin.us/5126 |
| 06:03.17 | emsLinux | i get the warning everytime Asterisk try to connecto to mysql trought ODBC |
| 06:03.32 | emsLinux | WARNING[2303]: app_voicemail.c:5095 inboxcount2: Failed to obtain database object for 'asterisk'! |
| 06:03.44 | emsLinux | i get that warning for every extension i have |
| 06:04.27 | *** join/#asterisk kleszcz (tick@80.54.23.253) |
| 06:05.00 | emsLinux | check the whole error message http://pastebin.us/5127 |
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| 06:07.23 | GreatSUN | re |
| 06:07.38 | kaldemar | emsLinux: have you configured odbcinst.ini and odbc.ini? |
| 06:08.13 | turtlefence | what happens if you 'voicemail show users' in the cli |
| 06:08.22 | emsLinux | yes, let me send them to you |
| 06:08.56 | kaldemar | do you have res_odbc.so loaded? "module show like odbc" will tell. |
| 06:09.26 | kaldemar | turtlefence: go try |
| 06:09.38 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
| 06:10.08 | emsLinux | http://pastebin.us/5128 |
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| 06:10.55 | emsLinux | Module Description Use Count |
| 06:10.56 | emsLinux | res_odbc.so ODBC resource 0 |
| 06:15.19 | emsLinux | any idea? |
| 06:17.40 | kaldemar | in voicemail.conf you should have odbcstorage=asterisk and a table defined with odbctable. does that table exist in your DB? |
| 06:18.22 | emsLinux | honestly, i dont know |
| 06:18.26 | emsLinux | let me check |
| 06:20.38 | cneb3000 | what a beautiful morning in the UK today :) |
| 06:21.15 | kaldemar | not raining too much? |
| 06:21.40 | cneb3000 | hmm.. just started.. good job n' all - i cycle to work :| |
| 06:22.37 | emsLinux | no i dont, i dont have that table in asterisk database |
| 06:22.41 | wdoekes2 | good morning |
| 06:22.58 | cneb3000 | and to you sir |
| 06:23.31 | kaldemar | emsLinux: there's your reason. |
| 06:24.08 | emsLinux | let me try ;) |
| 06:25.17 | emsLinux | im not very good with DBs, can u show me how to create correctly that table inside the asterisk database pls? |
| 06:26.08 | kaldemar | you'll find the table layout here: http://ofps.oreilly.com/titles/9780596517342/asterisk-DB.html#Database_id276920 |
| 06:26.36 | wdoekes2 | and the create table syntax here: http://dev.mysql.com/doc/refman/5.1/en/create-table.html |
| 06:27.25 | emsLinux | =( i don't know how to do it, mine are servers, not DBs |
| 06:27.46 | emsLinux | like this CREATE TABLE authors (id INT, name VARCHAR(20), email VARCHAR(20));? |
| 06:27.57 | wdoekes2 | yeap.. go try it |
| 06:30.12 | cneb3000 | emsLinux: it would be SOMETHING like CREATE TABLE voicemail (uniqueid INT, dir VARCHAR(80)) |
| 06:30.27 | cneb3000 | emsLinux: look into something called phpMyadmin |
| 06:30.41 | cneb3000 | emsLinux: It's an easy to use web based front end to MySQL |
| 06:31.06 | cneb3000 | in fact, it'll probably have a way of creating tables without running stuff like that ^^^ i dont know I've never really checked.. |
| 06:33.56 | emsLinux | is this the correct syntax? |
| 06:33.57 | emsLinux | CREATE TABLE voicemessages (uniqueid serial PRIMARY KEY,msgnum int4,dir varchar(80),context varchar(80),macrocontext varchar(80),callerid varchar(40),origtime varchar(40),duration varchar(20),mailboxuser varchar(80),mailboxuser varchar(80),recording lo,label varchar(30),"read" bool DEFAULT false,flag varchar(10)); |
| 06:35.10 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 06:35.14 | wdoekes2 | serial is not a datatype in mysql, use "uniqueid int auto_increment primary key" |
| 06:35.37 | wdoekes2 | or are you using postgres? |
| 06:36.23 | wdoekes2 | if you're using mysql, int4 and bool don't exist either, use int and tinyint(1) |
| 06:36.33 | emsLinux | mysql |
| 06:37.16 | wdoekes2 | and the 'lo' datatype I've never heard of.. I suspect you want BLOB |
| 06:37.24 | emsLinux | yes, is a blob |
| 06:37.38 | wdoekes2 | run the command and fix it until it gives no errors |
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| 06:44.04 | emsLinux | nop, keep getting the error at tinyint(0) |
| 06:44.41 | emsLinux | thank for your help guys, but i'm stupid for this DBs stuff |
| 06:45.24 | cneb3000 | emsLinux: Read what wdoekes2 said ^^^ 'tinyint(1)' |
| 06:45.39 | cneb3000 | emsLinux: Did you type tinyint(0) or tinyint(1) |
| 06:46.47 | wdoekes2 | (it would fail next on false.., but I'm not sure if emsLinux wants to be helped) |
| 06:47.13 | emsLinux | both |
| 06:47.48 | emsLinux | i tried both, i need it false, i read i have to use 0 for false |
| 06:47.56 | emsLinux | but it doesn't work either |
| 06:48.44 | cneb3000 | emsLinux: what's the error it's actually generated buddy? |
| 06:49.37 | emsLinux | You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near '"read" tinyint(0),flag varchar(10))' at line 1 |
| 06:50.19 | emsLinux | i need to create this table, http://ofps.oreilly.com/titles/9780596517342/asterisk-DB.html#Database_id276938, but i don't know anything about DBs |
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| 06:51.30 | wdoekes2 | emsLinux: you're trying ansi sql, you need funky mysql settings for that |
| 06:51.35 | wdoekes2 | try `read` instead of "read" |
| 06:52.39 | wdoekes2 | (which incidentally is written at note [a] of the first url) |
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| 06:53.25 | emsLinux | i thought it was telling me to read that line, lol |
| 06:53.37 | emsLinux | thank guys, it worked |
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| 06:54.23 | cneb3000 | emsLinux: Most useful lesson I was ever taught administering any type of software or server... "Google is your friend" :D |
| 06:55.00 | cneb3000 | but I'm glad it worked :P |
| 06:56.00 | emsLinux | i know, but is 2am here in Colombia, i've been working for about 20 hours now, and i dont know anything about DBs, srry to bother you, but thank you =) |
| 06:56.21 | cneb3000 | 2ma :| |
| 06:56.25 | cneb3000 | am* |
| 06:56.36 | cneb3000 | i'd very much be in bed by then :D |
| 06:58.37 | emsLinux | im too obsesive to do that |
| 06:58.50 | emsLinux | but i think i should, im still getting the same error |
| 06:59.02 | emsLinux | maybe i did something wrong with the table |
| 07:00.44 | wdoekes2 | probably your error is earlier than that |
| 07:00.53 | wdoekes2 | you need to ensure that your odbc link is working properly |
| 07:01.08 | wdoekes2 | see the isql command line tool and your res_odbc settings |
| 07:02.55 | emsLinux | ok, ill recheck everything now i have the table |
| 07:03.15 | emsLinux | i have to sleep a little, thank you for your help, good night |
| 07:03.20 | wdoekes2 | night |
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| 07:38.12 | bsdbandit | good morning all |
| 07:38.19 | cneb3000 | hi ho |
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| 08:25.09 | just187 | how can i login into the asterisk cli ? |
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| 08:45.11 | ChannelZ | asterisk -r |
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| 09:08.12 | just187 | thx :D |
| 09:09.13 | just187 | when i execute a sip show peer what argument does follow? the username? |
| 09:10.03 | cneb3000 | just187: A little tip.. if you type 'sip show peer' then just try and type a '?' charecter, it will show all the possible arguments. |
| 09:10.06 | cneb3000 | :) |
| 09:11.13 | just187 | hm true .. nice feature ... |
| 09:11.20 | WiretapSeven | you can thank Cisco for that one |
| 09:11.49 | WiretapSeven | though I would much rather have had the syntax 'show sip peers' than 'sip show peers' |
| 09:12.17 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 09:12.36 | just187 | ohm sip show peers is correct |
| 09:13.19 | just187 | but not usefull :-/ |
| 09:13.32 | cneb3000 | just187: What are you trying to do? |
| 09:13.36 | kaldemar | or hit tab instead of ?. |
| 09:13.51 | just187 | i set up a vicidial server ... and now i want to know if it registered |
| 09:14.04 | kaldemar | sip show registry |
| 09:14.11 | WiretapSeven | then sip show peers is certainly useful |
| 09:14.25 | just187 | says 0 SIP registrations. |
| 09:14.39 | just187 | but i can see the username when i execute a sip show peers |
| 09:14.42 | WiretapSeven | then it didn't register |
| 09:14.55 | kaldemar | registry shows outbout registrations. if it doesn't show any lines, then there's something wrong with your register statements in sip.conf. |
| 09:14.57 | WiretapSeven | sip show peers will show the peer entry, connected or not |
| 09:23.45 | *** join/#asterisk zamba (marius@flage.org) |
| 09:23.55 | zamba | i'm looking for a good conferencing phone for asterisk.. any suggestions? |
| 09:23.59 | zamba | a physical unit |
| 09:24.31 | just187 | snom 370 / 870 are pretty nice ... |
| 09:25.42 | zamba | not snom meetingpoint? |
| 09:27.49 | just187 | oh i guess i didnt get what you want .. you dont want to do conference calls .. you want all people in one room to do a conference call ? |
| 09:28.06 | just187 | all yelling into one box ? |
| 09:28.21 | zamba | just187: yup |
| 09:28.39 | just187 | i c .. one sec .. will check which one is in our meeting room .. it works fine :-P |
| 09:28.53 | zamba | ok, thanks :) |
| 09:28.58 | WiretapSeven | zamba, I've used the non-SIP polycom soundstations, love them, so the SIP ones should be just as good |
| 09:29.26 | just187 | and yes .. its a snom meetingpoint :D |
| 09:29.31 | zamba | just187: hehe |
| 09:29.37 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 09:29.39 | schmidts | hello |
| 09:29.42 | zamba | looking at polycom soundstation ip 5000 now |
| 09:30.04 | cneb3000 | Zamba: I set one of those up the other day. It was great at what it did! |
| 09:30.28 | zamba | cneb3000: ip 5000? |
| 09:30.36 | cneb3000 | zamba: yes sir |
| 09:30.41 | zamba | cool, cool |
| 09:31.03 | WiretapSeven | can't remember which one it was Cisco rebranded as the 7937 |
| 09:31.30 | cneb3000 | requirements of a good conference room phone... must not sound distorted at loud volumes.. mic must be built for a hands free environment |
| 09:31.38 | cneb3000 | the ip 5000 does both of those well :) |
| 09:31.44 | zamba | cool, cool |
| 09:31.48 | zamba | will look further into that |
| 09:31.53 | zamba | thanks for the input, guys |
| 09:32.01 | cneb3000 | no problemmo |
| 09:32.53 | WiretapSeven | glad to be of service |
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| 10:20.23 | kaii | hmm .. in libpri 1.2 there was an option in the makefile: "LAYER2ALWAYSUP" to disable power savings mode. now in 1.4, this option has vanished. does anyone know, how to keep the layer2 always up with libpri 1.4(.11.5) ? |
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| 10:25.24 | skrusty | morning |
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| 10:32.49 | kaldemar | kaii: that is a bristuff option that is not found in vanilla libpri 1.2 either. |
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| 11:26.47 | *** mode/#asterisk [+o file] by ChanServ |
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| 11:57.40 | MariusAgon | Hello, do you guys know, where I can find some info about Skills feature in Asterisk? |
| 12:01.21 | *** join/#asterisk fish-bulb (~qcstewart@nat/digium/x-sgdszlvjdrzotrot) |
| 12:03.19 | leifmadsen | MariusAgon: you have to build it yourself |
| 12:03.57 | MariusAgon | leifmadsen, I'm just trying to find some basics. |
| 12:04.06 | kaldemar | MariusAgon: http://ofps.oreilly.com/titles/9780596517342/asterisk-ACD.html |
| 12:04.24 | kaldemar | learn how the queues work first. |
| 12:05.08 | leifmadsen | I have one client I'm going to build this for on monday, and I'm going to take the approach of using multiple queues (one for each skill) then loading the agents into those queues and using penalties to set their primary and secondary skills |
| 12:05.51 | leifmadsen | the alternative is to use something like Local channels and then perform lookups in a database prior to calling the agent to determine if the skillset requested by the caller matches, and if not, to return Congestion() so the caller re-enters the queue |
| 12:09.47 | MariusAgon | i already know somethings about queue routing, and planing on implementig skill routing using local channels and stuf. but i trying to come up effective and simple to implement skil based routing strategies. my colegue says he saw announcement in mailing list abount 1.8 and skills? |
| 12:10.41 | *** join/#asterisk just187 (~just187@217.148.237.2) |
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| 12:12.53 | kaldemar | MariusAgon: it's all DIY. |
| 12:13.18 | MariusAgon | OK, thank you guys :) |
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| 12:22.59 | kaii | kaldemar: thx for pointing out that LAYER2ALWAYSUP is bristuff(ed) .. do you know how to achieve that behaviour with vanilla code? (libpri 1.4.11.5, dahdhi 2.4.1.2, asterisk 1.8.4) |
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| 12:30.15 | kaldemar | kaii: well, you can download bristuff and take a look at the patch for libpri. the changes for the option are quite minor but the patch is for libpri 1.4.3 and it can't be used as is for 1.4.11.5. |
| 12:30.18 | QuantumSchema | Hey all! I have a queue with "ringinuse=no" set. Is there a response I could provide in the dialplan when calling an agent that responds with "In Use" so it doesn't attempt to call the agent and doesn't generate a "RINGNOANSWER"? |
| 12:30.32 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 12:37.25 | kaldemar | kaii: q921.c seems to have gotten a rewrite since 1.4.3, so it won't be completely trivial to create a new patch. |
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| 13:07.20 | WIMPy | kaii: If you are able and willing to change to misdn2, it's configurable in LCR. |
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| 13:15.13 | jaytee | tons of new phishing emails claiming to be Skype updates or Adobe Reader updates floating around the interwebs today. |
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| 13:38.32 | pushpop | Anyone know a good deal about Aastra phones? |
| 13:43.28 | malcolmd | negative |
| 13:47.06 | otwieracz | Hello. |
| 13:47.14 | otwieracz | Anyone is using Asterisk with SIPGATE? |
| 13:47.38 | otwieracz | Im still getting: |
| 13:47.46 | *** join/#asterisk MRH2 (~chatzilla@62.49.242.3) |
| 13:47.48 | otwieracz | chan_sip.c:19639 handle_response_register: Failed to authenticate on REGISTER to '<my-sip-id>@sipgate.com' (Tries 3) |
| 13:47.59 | otwieracz | http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk |
| 13:48.06 | otwieracz | I followed this howto. |
| 13:48.41 | *** join/#asterisk coppice (~chatzilla@79.194.17.210.dyn.pacific.net.hk) |
| 13:49.14 | MRH2 | Anyone know why this doesnt; seem to work exten => _+.,1,Dial(SIP/${EXTEN:1}@something) |
| 13:49.22 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:49.39 | MRH2 | for numbers such as +44123456 |
| 13:49.59 | kaldemar | MRH2: how does it not work? |
| 13:50.04 | jaytee | MRH2, how do you dial a + from a phone? |
| 13:50.16 | MRH2 | you don;t have to use a phone |
| 13:50.23 | MRH2 | to dial |
| 13:50.31 | jaytee | nods |
| 13:53.36 | *** join/#asterisk Hail2theKingBaby (~john.dunn@194-176-87-181.onyx.net) |
| 13:54.24 | MRH2 | just get failed so falling back to exten 's' |
| 13:54.39 | MRH2 | was wondering if it doesn;t play well with the + sign |
| 13:55.00 | MRH2 | oh i have an idea |
| 13:55.06 | _Corey_ | MRH2: If your carrier sends you a +, you need to have it in the dialplan |
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| 13:55.37 | _Corey_ | like: exten => +12221112222,1,Answer |
| 13:55.38 | kaldemar | MRH2: show a CLI output of a call with verbosity enabled. |
| 13:57.18 | MRH2 | I think there might be another place this app is sanitising input (so as to just spit out numbers) way before it touches asterisk |
| 13:57.42 | MRH2 | i'll need to check again - thanks anyway |
| 14:10.07 | saxa | hi ppl , me again. |
| 14:10.12 | saxa | I get this |
| 14:10.15 | saxa | [May 27 16:29:56] NOTICE[25265]: chan_iax2.c:11870 __iax2_poke_noanswer: Peer 'brastrak' is now UNREACHABLE! Time: 330 -- Registered IAX2 'brastrak' (AUTHENTICATED) at 189.105.87.211:4569 |
| 14:10.19 | saxa | [May 27 16:36:52] NOTICE[25264]: chan_iax2.c:10894 socket_process: Host 189.105.87.211 failed to authenticate as brastrak |
| 14:10.22 | saxa | quadser*CLI> |
| 14:11.06 | saxa | so it authenticates, but when i do a call from my sip phone to the italy server it just appears that chan_iax2.c error |
| 14:11.15 | saxa | s/error/notice |
| 14:11.45 | saxa | with failed to auth. any ideas where to begin looking ? |
| 14:12.09 | saxa | both are 1.8.4.1 |
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| 14:49.08 | kaii | kaldemar, WIMPy: thx for your information regarding layer2alwaysup .. i'll go home and cry now. have a nice weekend. |
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| 15:00.37 | puzzled | hi |
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| 15:03.06 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:03.40 | blee | Hey all, having an issue with my asterisk box not responding to 200OK's my polycom phone is sending when trying to answer a call |
| 15:03.41 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:04.11 | blee | Wireshark seems to think that all the SIP is part of the same dialog, but asterisk ignores the 5+ 200OKs that get sent |
| 15:08.30 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-4.saucontech.com) |
| 15:09.44 | fullstop | Hi. I've never thought about this before, but has anyone here ever had the decimal portion of UNIQUEID roll? Mine's at 1306508937.2452042 now. |
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| 15:19.18 | *** join/#asterisk sereal (~jjrh@132.213.238.4) |
| 15:19.54 | sereal | is there anywhere i can download the asterisk documentation/wiki from the offical asterisk page? for some reason my browser is loading the asterisk pages incrediblly slow. |
| 15:20.18 | sereal | voip-info.org doesn't have the infomation I need. |
| 15:24.10 | citywok | sereal: doubtful |
| 15:30.14 | fullstop | sereal: google cache? |
| 15:30.28 | sereal | maybe, but thats a pain in the ass :p |
| 15:30.35 | fullstop | this is true |
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| 15:44.38 | MrNemus | so what would cause sip channels to stay open and have 0x0 (nothing) in there format |
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| 15:53.05 | davies147 | Hi, quick question about issues.asterisk.org if I may? |
| 15:53.28 | davies147 | I have a link to an attachment: https://issues.asterisk.org/file_download.php?file_id=26335&type=bug is there a way to find what bug it is attached to? |
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| 16:00.01 | sereal | for some reason the voicemail isn't reading the button pressed, like I can't hit 3 for advanced options and the lik |
| 16:00.04 | sereal | *like |
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| 16:31.51 | otwieracz | Is there any easy way to create my own voice-dtfm-menu with Asterisk? |
| 16:32.15 | otwieracz | (Press one for technical support, press 2 for sales, press blahblahblah) |
| 16:32.46 | leifmadsen | otwieracz: yes |
| 16:32.50 | leifmadsen | that's just dialplan |
| 16:33.01 | leifmadsen | primary application will be Background() |
| 16:33.02 | leifmadsen | ~thebook |
| 16:33.02 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 16:34.06 | otwieracz | http://www.voip-info.org/wiki/view/Asterisk+cmd+BackGround |
| 16:34.10 | otwieracz | leifmadsen: â |
| 16:34.11 | otwieracz | like this? |
| 16:34.18 | leifmadsen | otwieracz: sure |
| 16:34.46 | otwieracz | Is there any voice-files-repository? |
| 16:34.50 | otwieracz | I have ugly voice ;-) |
| 16:35.05 | fullstop | I wonder how long it will take for the digium asterisk wiki to start returning results above voip-info in google. |
| 16:35.07 | leifmadsen | asterisk comes with many sound files |
| 16:35.22 | leifmadsen | fullstop: probably as soon as you can read everything without logging in... |
| 16:35.25 | fullstop | otwieracz: you could always hire someone. Allison Smith has done fantastic work for us. |
| 16:35.36 | otwieracz | :) |
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| 16:41.01 | neurosys | is there a site for certified hardware (MBs, etc) for asterisk? |
| 16:42.10 | malcolmd | negative |
| 16:42.29 | malcolmd | most anything under the sun will work |
| 16:47.09 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 16:50.17 | Hail2theKingBaby | Does anyone have any experience in connecting a Mitel 5212 to Asterisk? |
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| 16:53.35 | paulc | Hail2theKingBaby: No but I'd love to try.. always liked Mitel gear :-) |
| 16:59.08 | sereal | people love that book |
| 16:59.21 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 16:59.26 | sereal | I found it was easier to just read the asterisk docs and ask google any questions |
| 16:59.39 | sereal | which usually are answered in voip-info.org |
| 17:02.50 | jaytee | and usually outdated |
| 17:04.16 | *** join/#asterisk gruvfunk (~chatzilla@user-160uac8.cable.mindspring.com) |
| 17:11.16 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 17:11.37 | fullstop | It is amusing to see notes in voip-info.org talking about recent changes, but they mean in Asterisk 1.2 |
| 17:12.04 | Freeaqingme | voipinfo is kinda unmaintained lately |
| 17:12.31 | coppice | fullstop: voip-info is almost useless, because nothing is date stamped |
| 17:17.00 | sereal | I have found most of the info useful. |
| 17:17.08 | sereal | <PROTECTED> |
| 17:26.06 | sereal | In the VoiceMailMain how do I specify the context with out the user? |
| 17:26.21 | sereal | just doing VoiceMailMain() puts you in the default context. |
| 17:27.02 | Corydon76-home | @whatever |
| 17:27.15 | sereal | so VoiceMailMain(@whatever) ? |
| 17:27.20 | Corydon76-home | yes |
| 17:27.57 | anonymouz666 | Corydon76-home: why is -home? and not -dig? |
| 17:28.27 | sereal | thanks, I thought I tried that but I must of not been putting the @ |
| 17:30.19 | Corydon76-home | anonymouz666: I'm no longer employed by Digium |
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| 17:52.52 | sereal | whats the difference between callgroup and pickupgroup? |
| 17:55.29 | wdoekes2 | the pickupgroup phones can pick up calls from callgroup phones |
| 17:56.13 | Corydon76-home | Think of callgroup as tier1 support and pickupgroup as the corresponding managers of tier1 support |
| 17:56.14 | sereal | so for phone1 to pickup phone2's calls they both need callgroup=1 and pickupgroup =1 ? |
| 17:56.59 | wdoekes2 | no |
| 17:57.09 | wdoekes2 | phone1 needs pickupgroup=1 |
| 17:57.10 | Corydon76-home | No, phone2 has a callgroup of N and phone1 has a pickupgroup of N |
| 17:57.29 | *** part/#asterisk gmaruzz (~gmaruzz@static-217-133-80-112.clienti.tiscali.it) |
| 17:57.47 | Corydon76-home | If both have the same callgroup and pickupgroup, then they can pickup each other's calls |
| 17:57.54 | sereal | okay thats what I want. |
| 17:58.13 | sereal | so you can pickup any phone calls from any phone in the office |
| 17:59.29 | sereal | so is there no way to have it as like *8+your extension for callpickup? |
| 18:01.44 | paulc | sereal: check out "core show application pickup" |
| 18:02.40 | sereal | er is that something with the AMI or documention? |
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| 18:10.11 | gruvfunk | greets all |
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| 18:14.29 | gruvfunk | I'm playing around with SMS() app and sending a text message from my mobile phone to my SIP DID I get: "Error invalid number. Please re-send using a valid 10 digit mobile number or valid short code." |
| 18:15.03 | gruvfunk | Using IPcomms free DID ringing into Asterisk |
| 18:15.36 | paulc | sereal: no, it's a CLI command - type "core show application pickup" at your *CLI> prompt and see what it says. Or check out https://wiki.asterisk.org/wiki/display/AST/Application_Pickup |
| 18:16.19 | paulc | gruvfunk: I thought the SMS app was for sending messages out? (and is your SIP DID definitely SMS-capable? who's it from?) |
| 18:16.29 | *** part/#asterisk [intra]lanman (~lanman@freeswitch/developer/intralanman) |
| 18:17.38 | gruvfunk | paulc: from IPcomms (unsure capability), and my read is you could send and receive (exchange) SMS text with compatible mobile phones, hard phones, and applications |
| 18:18.35 | paulc | gruvfunk: I think it's more for places that support SMS over analog FSK signalling - see https://wiki.asterisk.org/wiki/display/AST/Application_SMS |
| 18:21.49 | sereal | what happens with call pickup when two phones are ringing at the same time? does just the newest one get choosen? |
| 18:23.06 | paulc | I think the pickup app is more for "directed call pickup" - you punch an extension number in, the ringing phone you want to answer |
| 18:23.25 | *** join/#asterisk Faithful (~Faithful@202.189.73.144) |
| 18:23.45 | paulc | it kinda suggests that if you don't supply one, it does regular behaviour call pick up (which I'd interpret as "answer the longest ringing phone first", but I'm not 100% that's right - one to test and play with I guess) |
| 18:25.24 | gruvfunk | so is SMS over SIP a reality or not? anyone? |
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| 18:25.36 | sereal | I'll have to look at the Pickup application more, i'm using callpickup from features.conf and everyone can answer anyones calls |
| 18:26.00 | sereal | I'm not sure how you would use the Pickup() application to dial your extension and pick it up |
| 18:26.29 | sereal | unless I made like *extennumber the number you dial |
| 18:28.11 | *** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net) |
| 18:29.11 | paulc | sereal: yeah - exactly like that :) |
| 18:29.44 | paulc | you don't use pickup in features.conf and instead have something like exten => _*8xxx,1,Pickup(${EXTEN:2}) |
| 18:30.07 | paulc | Then when extension 223 is ringing, *8223 would pick it up from another phone |
| 18:30.17 | sereal | should I not use the pickup feature in features.conf? |
| 18:31.29 | sereal | or is just preference? |
| 18:36.48 | *** join/#asterisk cyphorious (~cyphoriou@chello062178189196.2.15.tuwien.teleweb.at) |
| 18:42.45 | Corydon76-home | sereal: different approaches from different contributors. the callgroup/pickupgroup is the original feature |
| 18:43.45 | gruvfunk | Generic question: Can I build an Asterisk based solution to accept SMS text messages from the public? (like in those shows where you send in your vote, I'm doing something different..) |
| 18:45.17 | *** join/#asterisk mKn0wt (~Taisigue@190.184.110.150) |
| 18:47.11 | paulc | gruvfunk: Not so much with Asterisk. But there are other ways of doing it - mostly with web services. Check out www.twilio.com or www.smsified.com |
| 18:47.14 | *** join/#asterisk DaveCanoe (~Dave@strike.eicat.ca) |
| 18:49.09 | WIMPy | gruvfunk: It might be possible via a PSTN gateway, but you really want a dedicated solution for that. |
| 18:51.22 | *** join/#asterisk Micc (~Micc@64.139.101.26) |
| 18:52.23 | Micc | is there a known bug in session timers? my provider is saying they are sending uac and asterisk is still trying to refresh the session. |
| 19:00.03 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:00.03 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 19:04.58 | sereal | when doing the CallPickup I get "ast_get_group: Ignoring invalid group" |
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| 19:05.31 | sereal | I'm doing PickUp(1234@context) |
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| 19:23.36 | SuperNull | Hey all im required to update some dialplan on 1.2 .. i have a bunch full digit extensions (non-regexp) but when they do not match i want it to fall through to another context .. am i able to just use exten => .,1,goto(context,xxx,1) ? |
| 19:25.04 | WIMPy | Use _X. |
| 19:25.14 | *** join/#asterisk gnuday (~gnuday@host86-163-111-44.range86-163.btcentralplus.com) |
| 19:25.28 | SuperNull | alright thanks been a while since i have had to do dialplan heh. |
| 19:25.38 | WIMPy | _. will also match the special extension which might have unwanted side effects. |
| 19:25.50 | SuperNull | alright probably true. |
| 19:26.11 | gnuday | Has anybody written an OCF resource agent for asterisk 1.6.x or 2.8.x ? Thanks |
| 19:27.30 | gnuday | Sorry I meant an OCF resource agent for asterisk 1.6.x or 1.8x |
| 19:28.56 | *** join/#asterisk JonathanRose (~jonathan@nat/digium/x-cbjixxlmvwplganh) |
| 19:29.12 | JonathanRose | .figlet figlet |
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| 19:31.37 | *** join/#asterisk Joe_CoT (~joecot@pdpc/supporter/active/joe-cot) |
| 19:31.47 | Joe_CoT | yo, anyone familiar with using gtalk with asterisk? |
| 19:32.04 | WIMPy | ~ask |
| 19:32.04 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:32.56 | Joe_CoT | k. I have gtalk.conf and jabber.conf set up in pretty much the same way as described in the asterisk docs, but I can't make outgoing calls. asterisk is completed with openssl support. |
| 19:33.21 | Joe_CoT | Whenever i make a call, the message in the logs is "no gtalk capable clients to talk to". This is with 1.8.4 |
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| 19:37.22 | Joe_CoT | oh, and apparently making a call to a phone number works, and causes asterisk to immediately shut down |
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| 19:55.37 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 19:58.02 | mKn0wt | hey guys any good billing system that works good on astrisk 1.4 |
| 19:58.04 | mKn0wt | ? |
| 20:00.38 | jaytee | A2Billing? |
| 20:00.53 | jaytee | Google is your "best buddy" |
| 20:01.45 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:01.45 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 20:07.27 | Joe_CoT | OK, still working on gtalk. so I can now make outgoing calls to phone numbers, and other people gtalk clients can call asterisk just fine. But I can't make outgoing calls. |
| 20:09.11 | mKn0wt | jaytee: yes i been working with A2B |
| 20:09.43 | mKn0wt | but iam having some problems |
| 20:10.47 | mKn0wt | thougt there were others cool option |
| 20:10.47 | mKn0wt | tank you bro |
| 20:14.48 | *** join/#asterisk gnuday (~gnuday@host86-163-111-44.range86-163.btcentralplus.com) |
| 20:16.51 | chazzam | ~thebook |
| 20:16.51 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
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| 20:32.37 | mKn0wt | \ |
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| 21:15.25 | c4t3l | |
| 21:15.32 | c4t3l | hello world |
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| 21:30.03 | atan | From the command line can I ring a SIP extension somehow? I just want to see if it's connected. |
| 21:31.10 | Joe_CoT | atan, you don't need to ring it to see it's connected. do sip show peers |
| 21:31.27 | kaldemar | console dial or channel originate |
| 21:31.57 | atan | Joe_CoT: devices show up in that list but I can't always ring them. Routers sometimes drop the connections, for example. |
| 21:32.39 | kaldemar | atan: are you familiar with qualify? |
| 21:33.08 | atan | kaldemar, somewhat but the question I am playing around with is simply how often the cellular device really needs to keep-alive. |
| 21:33.17 | atan | Default is 15 seconds but 30 seconds helps battery life. |
| 21:33.41 | atan | How do I dial from console? voip-info says cial [<extension>] but it says no such command |
| 21:33.54 | atan | Err. Need console driver. |
| 21:34.08 | atan | I assume the console driver is something I need to install when I compile? |
| 21:34.09 | kaldemar | channel originate |
| 21:34.32 | kaldemar | console dial etc. are provided by chan_oss, chan_alsa or chan_console. |
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| 21:35.04 | kaldemar | "dial extension..." is old syntax. |
| 21:35.23 | leifmadsen | channel originate ... |
| 21:35.36 | leifmadsen | nevermind you're dialing via the console |
| 21:36.36 | atan | Okay good to go now. Thank you :-) |
| 21:39.07 | Joe_CoT | OK, so now that leifmadsen's around, time to reask my question :D I set up gtalk. I can make outgoing phone number calls, and take incoming calls from gmail. I can't make outgoing calls to gmail, I just get "no gtalk capable clients to talk to". Any idea what's up? |
| 21:39.34 | leifmadsen | has no idea |
| 21:39.49 | Joe_CoT | then I am doomed :( |
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| 22:00.51 | sequencer | hi all, does anyone know where to find a good basic tutorial on AGI scripting ? |
| 22:03.02 | russellb | that was quick |
| 22:04.54 | paulc | That's what she said |
| 22:05.04 | WIMPy | Maybe he asked google in parallel. |
| 22:05.17 | russellb | paulc: zing! |
| 22:05.26 | paulc | ba-dum psshhhhh - thank you ladies and gentlemen, I'm here all week.. (it's one of those afternoons - is it home time yet?) |
| 22:05.35 | russellb | almost |
| 22:05.47 | russellb | it's loud music time here at least (since most people have already bailed) |
| 22:10.24 | dlyneswork | russellb, ah...you work at corp hq, not at home? |
| 22:10.26 | paulc | Haha.. we have a guy in the office who leaves slightly early on a Friday and always cranks one heavy metal track for 3 or 4 minutes before he leaves.. |
| 22:10.37 | russellb | dlyneswork: yup, most of the time anyway |
| 22:10.52 | paulc | slow day in the office today.. I'm pretty much done, just practicing some "presentee-ism" whilst thinking about other stuff |
| 22:11.00 | dlyneswork | russellb, well, I guess less distractions that way |
| 22:11.10 | russellb | i find home less distracting :-) |
| 22:11.33 | dlyneswork | russellb, oh, you must be referring to all the unnecessary meetings :o |
| 22:13.36 | russellb | heh, i do have a lot of meetings, but they are generally useful |
| 22:29.45 | *** join/#asterisk emsLinux (~emsLinux@190.251.111.118) |
| 22:31.04 | emsLinux | good afternoon guys, yesterday i was asking for help to connect asterisk with mysql db using ODBC, now i made ir, it appears connected, but is not recording the messages, can anyone help me? |
| 22:31.30 | emsLinux | this is the output from asterisk http://pastebin.us/5173 |
| 22:37.46 | emsLinux | does anyone know anything about voicemail storage with ODBC? |
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| 23:44.16 | carrar | moof |
| 23:44.42 | ChannelZ | FOOM! |
| 23:45.33 | paulc | FEE FI FO FUM? |
| 23:46.04 | ChannelZ | I smell the fart of an english mastif |
| 23:46.22 | WIMPy | Ooomph |
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| 23:48.50 | ChannelZ | Grrph, what happened to the little 'pop out' button on YouTube so I could watch 720p at 1:1 size? |
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