00:05.58 | *** join/#asterisk timahvo1 (~rogue@41.81.142.133) |
00:06.46 | *** join/#asterisk coppice (~chatzilla@host217-43-106-95.range217-43.btcentralplus.com) |
00:11.40 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
00:24.12 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
00:25.59 | SeRi | p3nguin: ah! ok I understand |
00:26.02 | SeRi | sorry for the afk |
00:26.57 | p3nguin | I think I have like five total translations, so it wasn't hard to narrow it down. |
00:28.02 | SeRi | lol |
00:28.07 | SeRi | sorry :P |
00:28.16 | p3nguin | Everything headed for 5060 is from 5060. |
00:28.46 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
00:28.56 | SeRi | ah I see. |
00:28.57 | AdamN | does asterisk 1.8 still use extensions.conf? |
00:29.05 | p3nguin | Only if you want it to. |
00:29.06 | SeRi | AdamN: yes |
00:29.38 | p3nguin | But out of curiosity, what else would you have expected? |
00:30.06 | AdamN | I just upgraded from 1.6, and it has lost all configuration, and running "config list" in the cli shows it is not seeing extensions.conf |
00:30.36 | p3nguin | Is pbx_config present? |
00:31.54 | AdamN | p3nguin: no |
00:32.01 | p3nguin | There's yer problem. |
00:32.32 | AdamN | p3nguin: within /etc/asterisk? |
00:32.44 | p3nguin | No |
00:33.02 | p3nguin | module show like pbx |
00:34.02 | AdamN | no, all module show displays is "res_adsi" |
00:34.12 | p3nguin | Then you have work to do. |
00:34.45 | p3nguin | In /etc/asterisk, what files do you have present and configured? |
00:35.31 | AdamN | well before I get into that I ran "module load PBX_config" |
00:35.38 | AdamN | and it is now loaded |
00:36.30 | p3nguin | And config list should show that pbx_config is using extensions.conf. |
00:36.39 | *** join/#asterisk [Outcast] (~anonymous@pool-96-252-45-211.bstnma.fios.verizon.net) |
00:36.43 | p3nguin | pbx_config /etc/asterisk/extensions.conf |
00:36.53 | AdamN | yes |
00:37.11 | p3nguin | Perhaps you have a jacked up modules.conf or something. |
00:38.43 | AdamN | where is modules.conf reside? |
00:38.56 | p3nguin | /etc/asterisk with all the other confs. |
00:39.23 | AdamN | that does not exist |
00:39.30 | AdamN | modules.conf that is |
00:39.34 | p3nguin | YOu need one. |
00:39.49 | p3nguin | You'll want to make sure you define autoload to be enabled. |
00:40.08 | p3nguin | Look at the sample modules.conf. |
00:40.26 | AdamN | I have a backup from before i updated this afternoon |
00:40.30 | p3nguin | Did you blow away all your confs from your previous setup? |
00:40.57 | p3nguin | I would have thought you'd leave them in place and attempt using them as they were. |
00:41.15 | p3nguin | There will be a few changes, but for the most part, things would have worked. |
00:41.23 | AdamN | nope, all the confs from /etc/asterisk were copied and backed up remote, the originals were left on the machine |
00:41.40 | p3nguin | So you're saying you didn't have a modules.conf before? |
00:42.09 | AdamN | sorry that got disjointed, I do have my old modules.conf, all the confs were backed up |
00:42.38 | p3nguin | I'd copy it into place and restart asterisk to see how it goes. |
00:42.46 | AdamN | ok |
00:43.50 | p3nguin | For the most part, things are the same in the confs. There are, however, a few subtle changes in some of them. I took my original confs and used them, then made the corrections for new values and changed syntax where needed. |
00:44.45 | p3nguin | voicemail.conf has a few changes, sip.conf changed a few parameters as well as added a bunch, some of the dial plan apps' syntax has changed slightly, et cetera. |
00:45.17 | AdamN | I notice alot of the cli commands are gone. |
00:45.28 | p3nguin | Nah, they aren't gone. |
00:45.38 | p3nguin | You're just looking in the wrong places. |
00:46.07 | AdamN | has o'rielly or the like updated for 1.8 yet? |
00:46.35 | p3nguin | For example, originate is now channel originate, soft hangup is now channel request hangup, and so on. If you take a look at your cli_aliases.conf, you can restore all of the old commands. |
00:46.42 | p3nguin | ~book |
00:46.42 | infobot | rumour has it, thebook is 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://www.asteriskdocs.org/ or ~buybook |
00:47.40 | AdamN | thank you very much, that would have been a loooong process |
00:49.45 | p3nguin | I use my cli aliases for when I forget I'm not on 1.4, but I try hard to use all the new commands. |
00:51.08 | AdamN | well it looks like everything but voicemail is back up |
00:51.36 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
00:57.03 | AdamN | all my voicemail boxes are stating this user can't accept more messages. |
00:58.18 | p3nguin | chown -R asterisk:asterisk /etc/asterisk /var/lib/asterisk /var/log/asterisk /var/spool/asterisk /var/run/asterisk |
00:58.35 | p3nguin | (assuming you run asterisk as user asterisk and group asterisk like a sane person) |
01:14.34 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
01:15.56 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
01:17.07 | *** join/#asterisk RiceCracker (~RiceCrack@59.152.236.158) |
01:19.52 | *** join/#asterisk xpot-mobile (~xpot@166-70-100-198.ip.xmission.com) |
01:27.58 | SeRi | p3nguin: everything ok so far? |
01:28.06 | *** join/#asterisk ariel_ (u3533@pdpc/supporter/active/abatista) |
01:28.15 | ariel_ | Hello everyone |
01:30.22 | ariel_ | quick question about queues and agents. This is on a asterisk 1.4.33.1 system. The polycom phones are logged in, but from time to time the when a call comes in, it will ring the phone, for one ring then go away, then come back, it sometimes goes to another agent, but most of the time it just waits about 5 sec then rings the same phone. Only way around this is for us to do a reload or for the ag |
01:30.22 | ariel_ | ents to log off wait about 30 sec then log back in. This is using remoteagent log in.... |
01:31.47 | *** join/#asterisk neurosys (~neurosys@c-67-191-66-234.hsd1.fl.comcast.net) |
01:34.13 | *** join/#asterisk s[X] (~mark@eth589.qld.adsl.internode.on.net) |
01:34.52 | *** join/#asterisk xpot-mobile (~xpot@166-70-100-198.ip.xmission.com) |
01:37.42 | *** join/#asterisk s[X] (~mark@eth589.qld.adsl.internode.on.net) |
01:44.42 | *** join/#asterisk s[X] (~mark@eth589.qld.adsl.internode.on.net) |
02:06.45 | SeRi | p3nguin: I have not get my password from freenum.org... should I email them? |
02:09.41 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
02:11.04 | SeRi | nevermind |
02:12.45 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
02:15.24 | SeRi | p3nguin: Is there anything special I need to do @ freenum.org? I am loged in in my account |
02:17.24 | p3nguin | You just need to configure some DNS stuff. |
02:17.50 | SeRi | on my server? |
02:19.41 | p3nguin | no |
02:19.58 | SeRi | ok |
02:20.13 | p3nguin | Sign in on freenum.org. |
02:20.28 | *** join/#asterisk mintos (~mvaliyav@115.241.41.116) |
02:20.30 | p3nguin | In the DNS settings, create your settings. |
02:20.40 | p3nguin | (it's not magical) |
02:21.16 | SeRi | just did. |
02:21.20 | SeRi | :) |
02:21.21 | p3nguin | After you do DNS settings, then configure anything in the ITAD settings that you want to configure. |
02:21.53 | *** join/#asterisk master_of_master (~master_of@p57B54457.dip.t-dialin.net) |
02:39.51 | p3nguin | How's that working out for you? |
02:40.27 | SeRi | Trying now |
02:41.32 | SeRi | p3nguin: so with the context you gave me I dial 012623*262? |
02:42.01 | p3nguin | 012 is the dialing prefix for ISN. |
02:42.29 | p3nguin | If you were trying to dial 623 @ 262, then that would be correct. |
02:43.11 | SeRi | in there site they have an echo test number 623*262 |
02:44.22 | SeRi | I dial 012623*262 and I get a fast bussy tone |
02:44.23 | p3nguin | It isn't echoing. |
02:44.28 | p3nguin | Yeah, it's broken. |
02:44.32 | SeRi | ah ok |
02:45.08 | p3nguin | I'll give you one to test. |
02:45.14 | SeRi | Thanks! :) |
02:46.14 | p3nguin | Seems to be working. |
02:46.15 | SeRi | hahahaha! |
02:46.21 | SeRi | rofl! |
02:46.24 | SeRi | nice |
02:46.53 | SeRi | lol cool |
02:47.26 | SeRi | for me to recive incoming calls do I just create a normal context? |
02:47.54 | SeRi | s/recive/receive/ |
02:48.39 | p3nguin | You already have a context. |
02:48.48 | p3nguin | Now you just need to have extensions. |
02:49.01 | SeRi | lol :) |
02:49.05 | SeRi | indeed |
02:49.13 | SeRi | 3 months in and still get it confused |
02:49.14 | p3nguin | Calls to you via ISN will end up in your misc_calls context. |
02:49.24 | SeRi | I see |
02:49.38 | SeRi | can you share an example? |
02:49.40 | p3nguin | I have an incoming extension prefix to be able to determine ISN from regular SIP calls. |
02:49.44 | p3nguin | One moment. |
02:49.51 | SeRi | Thank You. |
02:51.01 | p3nguin | Can you choose an arbitrary prefix for ISN inbound? This will be on every extension dialed in. |
02:51.13 | p3nguin | example, 123 |
02:51.22 | p3nguin | or 99 |
02:51.59 | SeRi | sure. |
02:52.04 | SeRi | 223 |
02:52.24 | p3nguin | And your internal extension number is what? |
02:52.31 | SeRi | 1003 |
02:52.40 | SeRi | my office ext^^ |
02:54.52 | p3nguin | http://pastebin.com/ywgYtjYn |
02:56.20 | SeRi | cool. one sec |
02:56.30 | p3nguin | So if I dial 2231003*yourITAD, it should hit your unauthorized calls context and reroute it to 1003 in your internal context. |
02:57.13 | p3nguin | And it prepares the caller ID so that all you have to do is press the Dial key on your phone. It would change my number into 012mynumber. |
03:00.00 | SeRi | awesome. :) |
03:00.08 | SeRi | can we test it? |
03:00.47 | p3nguin | Yes. |
03:00.49 | p3nguin | Shall I dial? |
03:00.55 | SeRi | Please :) |
03:01.45 | p3nguin | Didn't work. |
03:02.22 | SeRi | Mhhhhhh |
03:02.24 | SeRi | one sec |
03:03.59 | p3nguin | The DNS isn't good. |
03:04.05 | SeRi | yea I just saw that |
03:04.08 | p3nguin | Maybe it hasn't propagated. |
03:04.11 | SeRi | it has not propagated. |
03:04.16 | SeRi | +1 |
03:04.31 | SeRi | ill wait till tomorrow |
03:04.37 | SeRi | Thanks any ways :) |
03:04.50 | SeRi | By the way I pass my class! |
03:05.14 | SeRi | is going to party! |
03:05.18 | SeRi | lol |
03:05.19 | SeRi | :P |
03:05.28 | SeRi | I really thought I was going to fail :/ |
03:05.42 | SeRi | That was the hardest class ever! |
03:08.35 | SeRi | p3nguin: what headset/mic do you use with your computer? |
03:09.13 | p3nguin | It's some $3 crap I got from Hong Kong via ebay. |
03:09.57 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
03:10.05 | SeRi | nice. I am shopping for one and found a few 3.99 on newegg. |
03:11.03 | p3nguin | I also have a $20 Logitech headset that I picked up at CVS on their 75% off clearance rack, but I've never used it. |
03:11.26 | SeRi | I see. |
03:11.47 | SeRi | well I was looking at a mic and usem y current head set..... |
03:12.05 | SeRi | s/usem y/use my/ |
03:12.16 | SeRi | not sure yet |
03:12.35 | p3nguin | You're planning to use a soft phone instead of your hard phone? |
03:13.22 | SeRi | for my laptop down stairs yet |
03:13.26 | p3nguin | oh |
03:13.30 | SeRi | yes* |
03:13.32 | SeRi | :) |
03:14.05 | SeRi | my wife is starting to like the idea voip sof phones and such :) |
03:14.18 | SeRi | damn that was all fucked up |
03:14.49 | s[X] | hey all |
03:14.50 | SeRi | but any who yes... she is now using her cell to call my ext on my office. |
03:15.03 | *** join/#asterisk gajini (~root@61.12.17.170) |
03:15.05 | SeRi | uising scip |
03:15.13 | SeRi | s[X]: hola |
03:15.53 | s[X] | I wish iphone included Native SIP |
03:15.56 | s[X] | Would be so nice |
03:16.03 | SeRi | dont we all |
03:16.09 | s[X] | even JB dont do it |
03:16.25 | JerJer | acrobits wont work s[X]? |
03:17.07 | s[X] | its not native is it ? |
03:17.32 | p3nguin | It's not native, but it's a SIP phone app. |
03:17.45 | s[X] | I like Bria |
03:17.48 | SeRi | android 2.3 branch has native sip |
03:18.00 | s[X] | Nokias have native sip for ages |
03:18.21 | SeRi | Nokias are like mercedes though |
03:18.46 | SeRi | you have to give your first born child for one of there phones |
03:18.53 | p3nguin | I have a Nokia 918. |
03:18.59 | SeRi | s/tehre/their/ |
03:19.38 | SeRi | lol! keep it around p3nguin they are good phones |
03:20.04 | p3nguin | I don't think it's good for anything. |
03:20.41 | SeRi | When the world sprungs in chaos it would become a nice weapon! |
03:21.10 | SeRi | put it inside a sock and wave the sob around! |
03:21.17 | s[X] | I had an N95 then an N96 |
03:21.20 | s[X] | SHouldve kept the N95 |
03:21.25 | s[X] | made the N96 eat shit |
03:21.28 | SeRi | smack the firs fucker in the head thatc omes close to you |
03:21.52 | SeRi | I bet the phone would still be in one piece |
03:22.06 | SeRi | s[X]: I had a 770 and a 800 |
03:22.18 | SeRi | I rape them to no end and sold them. |
03:33.42 | SeRi | p3nguin: is resolving for me now on this side. |
03:33.57 | SeRi | the hostname that is |
03:41.43 | p3nguin | Still nothing here. |
03:42.17 | SeRi | ok thanks :) |
03:56.05 | p3nguin | Where did you come up with the idea that the DNS was able to resolve? |
04:01.38 | p3nguin | I'm querying the freenum nameservers directly, and there's no record for you. |
04:02.34 | p3nguin | NXDOMAIN |
04:02.50 | SeRi | well I was doing a look up on the domain it self |
04:03.01 | SeRi | I guess I was wrong |
04:03.23 | s[X] | im ashamed to admit i didnt completely understand linux permissions untill recently |
04:03.24 | SeRi | your doing a look up on freenum.... ok I see |
04:03.44 | s[X] | lol |
04:04.06 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
04:04.22 | p3nguin | Here's an example: dig @anyns.pch.net 3.2.1.404.freenum.org. NAPTR |
04:04.57 | p3nguin | Querying one of the freenum ns for ITAD 404 |
04:05.08 | p3nguin | I get an answer. |
04:05.19 | p3nguin | Yours, nonexistent. |
04:05.26 | SeRi | o I see |
04:06.14 | p3nguin | So maybe it's broken. |
04:06.23 | p3nguin | Or maybe you didn't fill out the DNS settings. |
04:06.27 | p3nguin | I have no idea. |
04:06.30 | SeRi | I did.... |
04:17.44 | SeRi | p3nguin: any other ideas? |
04:17.58 | p3nguin | Contact admin@freenum.org about it. |
04:18.22 | SeRi | ok |
04:18.57 | p3nguin | Tell them exactly what you told me about things not working correctly. |
04:21.45 | SeRi | done. |
04:22.10 | s[X] | Whats the deal with ISN |
04:22.18 | s[X] | How does it work |
04:23.55 | p3nguin | It's a SIP URI which is obscured with a fancy DNS lookup. |
04:24.47 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
04:25.02 | p3nguin | The usual SIP URI is something like 123@my.host.com or name@my.host.com, but it's hard to dial those from a standard phone keypad. |
04:25.16 | p3nguin | So we use an ITAD instead of a host name. |
04:25.19 | p3nguin | ~itad |
04:25.20 | infobot | extra, extra, read all about it, itad is an IP Telephony Administrative Domain. Similar in nature to an AS (Autonomous System) number, it is administered by IANA (Internet Assigned Numbers Authority). An ITAD number is part of the TRIP specification in RFC 3219. Although TRIP never took off, ITAD numbers are being used by Freenum.org as part of an ISN (ITAD Subscriber Number). For more information about ISN numbers, see 'isn'. |
04:25.28 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
04:25.35 | p3nguin | And we use a * instead of an @. |
04:25.59 | p3nguin | So to call 123 on ITAD 404, you'd dial 123*404 from your regular keypad. |
04:26.32 | p3nguin | It makes SIP-SIP calling possible for more people. |
04:26.37 | s[X] | ah ok |
04:27.27 | s[X] | so you would register a ITAD point it to a subdomain, that points to ur asterisk box and bobs ur uncle ? |
04:27.46 | *** join/#asterisk ilj (~ilj@sourcemage/grimoire/apprentice/ilj) |
04:27.52 | p3nguin | Something like that. |
04:28.01 | s[X] | Sounds cool |
04:28.06 | s[X] | whats an ISN worth ? |
04:28.24 | hardwire | I need to get into freenum |
04:28.27 | p3nguin | It costs nothing, so I don't know how to calculate the value of it. |
04:28.43 | s[X] | Priceless ? |
04:28.58 | p3nguin | or worthless? |
04:29.27 | p3nguin | I'm suddenly reminded of the Family Guy episode where Carter is trying to pay off Peter to leave Lois alone. |
04:29.43 | SeRi | lol |
04:29.53 | p3nguin | She may be worth a million dollars to you, but to me, she's worthless! |
04:30.00 | s[X] | lol |
04:32.26 | p3nguin | No one ever calls me by ISN, but it's still a useful tool to have. Since it's free, you may as well go ahead and register for an ITAD so you can play with it. |
04:32.40 | s[X] | yeah tis what im thinking |
04:33.22 | s[X] | Just because im trying to understand it a little better, If i were to dial your iSN from my Soft phone |
04:33.35 | s[X] | Does it route direct asterisk box to asterisk box |
04:33.42 | s[X] | assuming u had an asterisk box |
04:33.49 | p3nguin | yes |
04:33.53 | p3nguin | That is exactly what it does. |
04:33.55 | s[X] | Thats pretty cool |
04:33.59 | s[X] | i want one |
04:34.00 | s[X] | lol |
04:34.05 | p3nguin | It's a SIP URI hidden with fancy DNS. |
04:34.12 | s[X] | yeah |
04:34.35 | s[X] | goes off to setup a subdomain for his ISN |
04:34.59 | s[X] | mmm which domain to pick from |
04:35.00 | s[X] | lol |
04:35.21 | SeRi | penusinyourface.org? |
04:35.29 | s[X] | i dont have that one |
04:35.33 | s[X] | i have penus-in-you-face.org |
04:35.45 | s[X] | or my-bum-burns-mommy.org |
04:35.54 | SeRi | lol j/k :) It is avail though! lol |
04:36.00 | s[X] | hahah |
04:36.25 | SeRi | shitshaper.com |
04:36.29 | p3nguin | heh |
04:36.30 | SeRi | dijib: ^^ |
04:36.48 | p3nguin | Keep in mind that no one dialing by ISN ever sees the domain name. |
04:37.05 | s[X] | yeah i know but i figured id use a domain that i likely wont retard |
04:38.34 | SeRi | fucking weather.com cut off free weather api. fuckers |
04:39.19 | SeRi | now I have to go dig a script for conky to work with conkyforecast |
04:39.20 | s[X] | Seri... |
04:39.24 | s[X] | BOB ? |
04:39.51 | SeRi | bob? |
04:40.14 | s[X] | Seiri, Inc. |
04:40.32 | s[X] | There is a ITAD registration with that as the Organization u can see how i jumped to the conclusion lol |
04:40.44 | SeRi | lol |
04:40.52 | SeRi | not me sr :) |
04:40.58 | s[X] | lol |
04:41.07 | s[X] | I figured as much just thought it was conincidence |
04:41.49 | SeRi | I am in there just not as bob :) |
04:43.05 | SeRi | p3nguin: some fucked up shit just happen :/ |
04:43.08 | p3nguin | I'm there as well, and also not "bob." |
04:43.15 | p3nguin | modem rebooted? |
04:43.25 | SeRi | my record @ freenum record got deleted... |
04:43.40 | SeRi | my freenum record* |
04:44.40 | s[X] | just because im more curious |
04:44.43 | SeRi | I wonder if they are working on it |
04:45.27 | s[X] | if ias assigned ITAD 123 |
04:45.29 | s[X] | was* |
04:45.34 | s[X] | and my Extension was 201 |
04:45.39 | s[X] | would all you need to dial is 201*123 |
04:45.49 | p3nguin | That's the idea. |
04:45.57 | s[X] | that seems overly simple, why on earth are there so few registrants |
04:46.15 | p3nguin | You have to have dial plan to do it, but that part is very easy as well. |
04:46.48 | s[X] | Woot, Submitted |
04:46.53 | s[X] | I await there contact |
04:47.52 | s[X] | When will telcos allow ISN dialing from land lines ? |
04:48.20 | SeRi | p3nguin: I sent you a msg with the record |
04:51.31 | SeRi | p3nguin: the context for inbound goes in misc calls right? |
04:59.15 | p3nguin | ISN calls will be anonymous calls. Anonymous SIP calls go to the context you have assigned in the general section of sip.conf. |
04:59.40 | SeRi | Just wanted to make sure. |
05:02.01 | *** join/#asterisk ChannelZ (channelz@burner.com) |
05:08.50 | *** join/#asterisk irroot (~gregory@197.110.156.226) |
05:11.45 | s[X] | mmm i wonder if i can make outbound calls on my samsung pbx to isn |
05:12.15 | p3nguin | If it does SIP, then you probably can. |
05:12.34 | s[X] | yes it does sip |
05:12.37 | s[X] | its a 7200s |
05:12.58 | s[X] | I will probably have to setup a outbound router for ISN |
05:13.11 | s[X] | ill email samsung and get them to do it remotely, i dont want to break anything |
05:13.29 | irroot | s[X] samsung 7200 is nice but SIP is extra $$$ and does NOT do nat properly |
05:13.46 | irroot | morning folks |
05:13.49 | s[X] | its a shitty SIP box imho |
05:13.53 | s[X] | morning irroot |
05:14.34 | irroot | s[x] the 7200 has a identity crisis does not know what it is does legacy PBX + VOIP + Ethernet switch + .... |
05:14.44 | SeRi | lol |
05:15.00 | SeRi | p3nguin: I am wondering if freenum does not like cnames... |
05:15.28 | p3nguin | I doubt it matters. |
05:16.10 | s[X] | Its very annoying because when i was doing my research I thought it was quite a SIP capable unit |
05:16.10 | *** join/#asterisk kikohnl (~kotis@udp019436uds.hawaiiantel.net) |
05:16.25 | s[X] | I got it recommended by 4 seperate companies who quoted me |
05:20.19 | s[X] | Samsung are apparently released quite a major overhauled version early next year to acoomodate for Multiple SIP Carriers & More control over codecs |
05:20.28 | s[X] | s/released/releasing |
05:21.11 | irroot | <PROTECTED> |
05:22.34 | irroot | the way we accomodate these products is pop in a small Atom based micro PC with asterisk on it let asterisk do the real work |
05:23.23 | s[X] | So instead of getting SIP trunks for the PBX just get SIP channels for the Asterisk box ? |
05:23.49 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
05:23.56 | irroot | s[X] and then get the PBX to "trunk" to asterisk |
05:23.57 | SeRi | s[X]: This is called "What they say and What they mean" http://pastebin.com/raw.php?i=4Pys0PMV |
05:24.39 | s[X] | mmm |
05:25.02 | s[X] | Because i have 8 SIP trunks for my SIP provider |
05:25.06 | s[X] | i could use those on asterisk couldnt i |
05:25.18 | p3nguin | ~siptrunk |
05:25.18 | infobot | siptrunk is, like, something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
05:25.35 | p3nguin | ~itsp |
05:25.35 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
05:26.11 | s[X] | thanks p3nguin |
05:27.03 | s[X] | slides the word channels in |
05:28.13 | irroot | p3nguin elephants have trunks and sip through them ??? is that a sip trunk |
05:28.28 | p3nguin | http://imagebin.org/185866 |
05:28.30 | SeRi | lmao |
05:28.52 | p3nguin | asterisk trunk |
05:30.34 | irroot | the term trunk is synonyms with a trunk line its not only in asterisk one of the more vocal ISP's advertise VOIP Trunk to replace existing analogue trunk .... confusing the masses |
05:30.38 | s[X] | frantically needs to photoshop a trunk onto the asterisx character |
05:31.17 | irroot | s[X] you know those Mweb fools here in ZA |
05:31.23 | s[X] | yes |
05:31.30 | SeRi | heads out to bed. ftw! |
05:31.38 | s[X] | later SeRi |
05:31.38 | p3nguin | Pill? |
05:31.42 | SeRi | Yes Sr! |
05:31.47 | p3nguin | :) |
05:31.51 | SeRi | lol |
05:31.52 | p3nguin | laytor |
05:31.56 | SeRi | Its amazing |
05:31.59 | irroot | SeRi cheers /me just woken up |
05:32.03 | SeRi | l8trs! |
05:32.05 | p3nguin | Enjoy the sleeps. |
05:32.11 | SeRi | wel g/m and g/n! |
05:32.14 | SeRi | :D |
05:32.16 | SeRi | cya guys |
05:32.25 | irroot | cheers |
05:33.11 | SeRi|zzZZzz | cya. |
05:33.44 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
05:38.09 | s[X] | irroot, im intrigued by setting up a asterisk box between my pbx and ITSP |
05:38.48 | irroot | s[X] we do it all the time sometimes even use the PRI / BRI interface to the PBX on asteriks |
05:39.27 | s[X] | Being that i have 8 licenses for PBX |
05:39.30 | s[X] | SIP* |
05:39.30 | irroot | we do a Inline where put a 2 port PRI in asterisk on port to the PBX one to Telco and the ITSP |
05:40.09 | *** join/#asterisk shido6 (~shido6@c-98-234-181-35.hsd1.ca.comcast.net) |
05:40.35 | irroot | then asterisk does the call logging via monitor the ACD via app_Queue and the ivr also add things like call limits by value so say 100R calls per month |
05:41.10 | irroot | also does a good job as TMS |
05:41.19 | s[X] | I wrote my own SMDR capturing app in PHP for the 7200s |
05:41.36 | s[X] | so I could capture call logs |
05:42.01 | irroot | also allows voip phones on legacy system |
05:44.35 | s[X] | Listens in on a port, grabs incomming stream. Cleans it up and stores it in an SQL Database |
05:45.41 | [TK]D-Fender | I did something similar in Turbo Pascal for a Vantage 25 system about 17 years ago :) |
05:45.54 | s[X] | :P im still learning |
05:46.08 | [TK]D-Fender | SMDR logging, feature codes for blacklisting, CID integration, etc |
05:46.21 | [TK]D-Fender | I had way too much free time |
05:48.16 | irroot | [TK]D-Fender morning there |
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07:35.53 | schmidts | good morning |
07:38.37 | james_zhu | :-D |
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07:43.14 | wdoekes2 | morning |
07:43.30 | dym | hi |
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08:00.53 | kikohnl | ~itsplist-us |
08:00.53 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
08:01.55 | sawgood | Is there a problem using numbers or symbols in a IAX2 context? |
08:03.56 | sawgood | I keep getting, "Registration Refused" when trying to setup an IAX2 trunk between 3 servers (any tips) |
08:04.51 | *** join/#asterisk phpboy (~shane@blowfish.x86.co.za) |
08:04.52 | kaldemar | check username/matching peer and secret. |
08:05.33 | phpboy | My asterisk keeps crashing and I can't for the life of me figure out why, looks like a memory related issue |
08:05.49 | phpboy | anybody got any ideas what I should put of my checklist of things to check? |
08:07.37 | kaldemar | ~debug |
08:07.37 | infobot | ACTION DeBuggers $1 |
08:07.50 | kaldemar | ~backtrace |
08:07.50 | infobot | backtrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt). See also https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
08:09.57 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
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08:11.50 | phpboy | where would the core files generally be stored? |
08:12.02 | phpboy | CentOS 5.5 |
08:13.25 | sawgood | something else other than username/secret is stopping IAX2 from working (I've been chekcing the context for correctness for 2+ hours) |
08:15.19 | phpboy | This does suck, the daemon doesn't crash, I can still get onto the console, it just stops accpeting calls |
08:15.24 | phpboy | goes completely quiet :T |
08:15.33 | ChannelZ | are you sure they are matching the right peers? |
08:15.45 | kaldemar | sawgood: pastebin your config and a CLI output of a registration attempt on the receiving side with iax2 debug and verbosity. |
08:15.54 | sawgood | I will pastebin shortly |
08:15.56 | singler | phpboy: then it probably deadlocks, not crashes, so you will not have core dump |
08:16.15 | singler | in that backtrace link is info about deadlocks too |
08:16.19 | kaldemar | phpboy: then see the deadlock section for the same page. |
08:16.30 | kaldemar | phpboy: which version are you using? |
08:17.13 | phpboy | 1.8.7.1 |
08:17.28 | phpboy | on a 64bit server if that makes any diffs |
08:19.03 | phpboy | this looks like it's going to be an issue on a production server :T |
08:19.55 | singler | deadlocking is already an issue on production, to take debug before killing it is not a big deal |
08:20.15 | phpboy | lol, good point |
08:20.25 | phpboy | "Instead take the 5 mins while everyone is freaking out to attach gdb to the running asterisk process and do" |
08:20.29 | phpboy | I'm going to do that |
08:20.49 | singler | it will be like 1 min if you practice before and prepare commands :) |
08:20.58 | singler | or even less |
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08:24.40 | sawgood | If I've made a stupid rookie style mistake, I am sorry in advance (its been several hours) |
08:24.44 | sawgood | http://pastebin.com/AiUwgmcS |
08:25.42 | sawgood | the devices are on the same Ethernet switch (with iptables off) |
08:25.58 | sawgood | there might be a 'cheap' router in front of the switch for Internet access |
08:28.06 | singler | I think host must be dynamic for it to register |
08:28.22 | kaldemar | sawgood: ^^^ |
08:28.41 | sawgood | I am trying to grab a CLI of the failed registration attempt |
08:28.50 | sawgood | ty |
08:29.25 | ChannelZ | yeah if the IPs are static there's no need to register |
08:29.40 | singler | sawgood: also if IP does not change, remove register line |
08:29.52 | sawgood | gone |
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08:30.11 | sawgood | how come after running iax2 reload (it takes a while for the client to re-register with the server) |
08:30.37 | kaldemar | sawgood: and change those credentials ASAP if they were real. |
08:31.11 | singler | sawgood: did you set host=dynamic or did you remove register line? |
08:31.19 | zyphlar | that's a pretty bad secret O_o use random.org please |
08:31.33 | sawgood | I removed the register line and changed the host=dynamic |
08:31.39 | ChannelZ | no |
08:31.40 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
08:31.55 | ChannelZ | host=dynamic to be used with register |
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08:32.08 | sawgood | ok register comment going back in |
08:32.10 | ChannelZ | If the machine's IPs are static, there's no need to use register, set host= to whatever |
08:32.38 | ChannelZ | the only thing register does is tell the remote end "here I am at this IP address" |
08:32.49 | ChannelZ | It's totally unrelated to whether or not a call will even work |
08:32.58 | sawgood | cool ... got it |
08:33.05 | ChannelZ | You can successfully register but still have bothced your peer configs and nothing will work |
08:33.16 | singler | sawgood: I guess you want to use qualify=yes to see if host is online |
08:33.35 | sawgood | I have removed the register line ... and I have host= static IP address |
08:34.06 | *** part/#asterisk james_zhu (~Administr@183.16.88.158) |
08:34.09 | ChannelZ | cross fingers and make a call over it :) |
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08:35.43 | qakhan | i want to setup speech to text on asterisk |
08:35.57 | qakhan | plz help me how to do that |
08:41.01 | sawgood | When I attempt to make a call, I can see on the client machine an error msg (saying NO Authority found) ... and on the server side I can see the call arriving but an error message saying the call cannot be delivered |
08:41.23 | singler | sawgood: can you pastebin cli logs? |
08:44.33 | sawgood | http://pastebin.com/HeTDfYzC |
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08:46.48 | kaldemar | sawgood: IAX2/173.13.158.17/501 is lacking authentication information. if you dial by ip address, use IAX2/username:secret@173.13.158.17/501. if you have correctly defined peers in iax.conf, use IAX2/IAX-ITSP/501. |
08:47.56 | sawgood | I tried using the IAX-ITSP context name in extensions.conf, but the call would not make it that far (I will try again) |
08:48.42 | sawgood | when you saying 'lacking authentication information' (is that the client or server side)? |
08:49.37 | sawgood | exten => 501,1,Dial(IAX2/IAX-ITSP/${EXTEN}) |
08:49.43 | sawgood | This is my statement in extensions.conf |
08:50.12 | phpboy | while I'm waiting patiently for asterisk to stop taking calls |
08:50.28 | phpboy | can anybody recommend something I an look into while it's running |
08:50.45 | phpboy | calls are cutting and all kinds of weird stuff at the mo :T |
08:51.10 | defswork | anyone any ideas why chanspy would drop the spier after a minute or so ? |
08:51.10 | singler | sawgood: what is cli output with that exten config? in pastebined log exten is using IP address |
08:51.20 | *** join/#asterisk like_a_horse (~like_a_ho@firect.saao.ac.za) |
08:51.49 | phpboy | 640 files in my /fd dir |
08:51.50 | phpboy | hmmmm |
08:52.39 | like_a_horse | hi all, i have a patton smartnode that doesn't seem to want to push calls through to my asterisk box. This was working on another asterisk box so the config on the smartnode seems to be ok but I still not seeing anything at all coming through in the rasterisk console |
08:53.06 | like_a_horse | a tcpdump show traffic to the asterisk box from the smartnode so i know its attempting a connection |
08:53.20 | like_a_horse | but i dont see anything in the rasterisk console |
08:53.46 | like_a_horse | is there a log file or something i can enable in the rasterisk console to output all sip attempts and tell me why they failed? |
08:56.57 | defswork | turn on sip debug |
08:58.43 | like_a_horse | core set debug .. ? |
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09:05.14 | qakhan | <PROTECTED> |
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09:57.23 | IsUp | hi |
10:12.26 | kaldemar | like_a_horse: "core set verbosity 10" and "sip set debug on" |
10:16.14 | like_a_horse | kaldemar, thanks.. |
10:24.26 | qakhan | hi, i want to setup sphinx on asterisk. please help me how to do that |
10:33.07 | like_a_horse | qakhan, on a asterisk server? |
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10:34.50 | like_a_horse | qakhan, logistically its not straight forward. Unless you live in Egypt. You realize setting up a sphinx on your asterisk box will most probably squash the server chassis to bits. Really not worth IMHO |
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10:46.49 | qakhan | like_a_horse i didnt get u |
10:48.55 | qakhan | sphinx is speech to text application |
10:49.09 | qakhan | i want to setup om asterisk |
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10:58.37 | Cadey | Hi guys, is there a current list of SIP related RFC's the latest release of asterisk supports/ad-hears to ? |
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11:44.54 | *** join/#asterisk gordonjcp (~gordonjcp@aramaki.gjcp.net) |
11:44.56 | gordonjcp | morning |
11:45.22 | gordonjcp | is anyone here using Cisco 7910 phones with asterisk, and are you prepared to comment on how good or bad they are? |
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12:07.35 | s[X] | gordon |
12:07.35 | s[X] | hey |
12:08.43 | s[X] | s/gordon/gordonjcp |
12:10.50 | gordonjcp | s[X]: hi |
12:11.11 | gordonjcp | incidentally, it's too late now to tell me that 7910G+SW phones are crap, I just bought two ;-) |
12:11.19 | s[X] | lol |
12:11.32 | s[X] | They actually dont work |
12:11.42 | s[X] | There is a bug with that specific model and asterisk |
12:11.45 | gordonjcp | I figured that since I'd got hold of the firmware and configured chan_sccp to the point that I can at least see its settings in the asterisk console, I can't go far wrong |
12:11.57 | s[X] | No im just messing with ya |
12:12.16 | gordonjcp | :-p |
12:12.27 | gordonjcp | for 20 quid I can't really go wrong |
12:12.33 | s[X] | I was actually going to ask how well they work as i wanna get myself a pair of cisco phones |
12:12.51 | s[X] | but i was thinking the 7941 |
12:12.57 | gordonjcp | well a couple of friends of mine are running some of the fancier cisco phones |
12:12.57 | gordonjcp | yeah |
12:13.18 | s[X] | I have found a place that has them for $99 AUS |
12:13.22 | s[X] | brand new |
12:13.29 | gordonjcp | nice |
12:13.52 | gordonjcp | that's insane, the cheapest I've seen them here is around £100 |
12:14.03 | qakhan | hi, i want to setup sphinx on asterisk. please help me how to do that |
12:14.11 | gordonjcp | AU$99 is about £63 |
12:14.14 | s[X] | yeah |
12:14.21 | s[X] | its insanely cheap |
12:14.24 | gordonjcp | I spent more than that on a curry |
12:14.30 | s[X] | http://www.computeralliance.com.au/parts.aspx?qrySearch=7941 |
12:15.11 | s[X] | only thing i need to check with them is its a 7941 not a 7940 |
12:16.43 | gordonjcp | oh no! |
12:16.48 | gordonjcp | it doesn't have speakerphone! |
12:16.55 | gordonjcp | that's okay, I bloody *hate* speakerphone |
12:16.56 | s[X] | yours ? |
12:17.05 | *** join/#asterisk elred_ (~elred_@unaffiliated/elred-/x-5010831) |
12:17.15 | s[X] | i never use speakerphone |
12:17.21 | s[X] | im in a shared office so i guess thats why |
12:18.51 | gordonjcp | I just don't like how they sound |
12:26.09 | s[X] | SPA3102 seems to be a deceny way to get a single FXO and FXS into asterisk |
12:28.00 | s[X] | and by decent i mean cheap |
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12:36.24 | gordonjcp | nice |
12:36.29 | gordonjcp | I've got an spa2100 |
12:37.32 | gordonjcp | dual fxs |
12:38.51 | s[X] | no fXO ? |
12:39.24 | gordonjcp | no |
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12:43.09 | danjekins | Hi, I wondered if anyone could point me in the right direction to fix the safe_asterisk issue I'm having. asterisk will start fine when just running asterisk, but if i run amportal start or service asterisk start etc then safe_asterisk will give me an error 127 |
12:43.39 | gordonjcp | can you still use faxmodem cards as FXOs? |
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13:00.29 | diegocn | hello ppl... how can i do a 'sip reload' from my linux console and not from asterisk console? |
13:01.02 | danjekins | you should be able to do asterisk -rx "sip reload" |
13:01.24 | danjekins | i think |
13:01.26 | s[X] | yeah |
13:01.40 | s[X] | spot on |
13:02.02 | diegocn | thanks danjekins |
13:02.20 | danjekins | youre welcome |
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13:33.27 | patrickod | I'm having trouble bridging calls betwen two trunks |
13:33.49 | patrickod | I've set up call forwarding, and the call dials outbound on the second trunk yet when they connect there is no audio being passed |
13:34.50 | tompaw | Guys, I have func_curl selected in menuconfig, it's compiled properly, yet asterisk says no function curl available. |
13:34.58 | tompaw | Do I have to enable it somehow? |
13:35.37 | tompaw | Aaah, nvm, my bad. |
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13:48.34 | kaldemar | patrickod: what technology are you using? SIP? |
13:52.46 | patrickod | kaldemar: yep |
13:53.14 | patrickod | I can see the calls progressing through a dialplan, they answer correctly etc |
13:53.27 | Diffen | Hello all. Is it possible to get my asterisk to look in the From header when receving an invite from the pstn? My invite looks like this: http://pastebin.com/CWdqAHDc |
13:53.31 | patrickod | but when it Packet2Packet bridges them neither side gets any audio |
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13:54.54 | kaldemar | patrickod: that's the fault. set directmedia=no under the peers in sip.conf. |
13:55.26 | patrickod | kaldemar: ok. |
13:58.21 | *** join/#asterisk irroot (~gregory@196-210-222-7.dynamic.isadsl.co.za) |
13:59.16 | patrickod | kaldemar: I've enabled that option on both peers but yet it's still Packet2Packet bridging the two |
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14:00.48 | [TK]D-Fender | Diffen, "core show function SIP_HEADER" |
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14:06.04 | patrickod | kaldemar: now it's native bridging but still no audio being received |
14:07.09 | JerJer | meep meep |
14:07.33 | Diffen | [TK]D-Fender: like this page shows. http://www.the-asterisk-book.com/unstable/funktionen-sip_header.html |
14:08.39 | [TK]D-Fender | Diffen, Amazing, isn't it? |
14:08.49 | Diffen | :D |
14:09.20 | Diffen | Its hard to look for something when you dont know the name of the thing you are looking for :D |
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14:12.36 | *** join/#asterisk FinboySlick (~shark@74.117.40.10) |
14:15.22 | FinboySlick | Anyone knows of a business-oriented scanner that just pretends to be a fax machine for dumb people who still expect to punch in a few numbers and hit send in a voip infrastructure? |
14:15.42 | FinboySlick | (obviously, it has to be asterisk-friendly) |
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14:21.46 | tzafrir | FinboySlick, mailing the result is good enough? |
14:22.06 | kaldemar | Diffen: that's why there are "core show applications" and "core show functions" to list them all. |
14:22.12 | tzafrir | But where do you define the address? |
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14:47.59 | FinboySlick | Hmm, netsplit... Anyway. Are fxs ATAs good enough with T38 now that one can just assume you plug in the fax machine set things up and it'll work? |
14:52.03 | r0m|u | FinboySlick, Thats a vague question..... there is so many ata's out there that one does not know which one supports T.38.... |
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14:54.52 | r0m|u | The one I know that support T.38 out of the box will be the new Cisco SPA-112 |
14:55.33 | FinboySlick | r0m|u: Well, it was more of a general question. Client is asking: "will my fax machine work" and expects a yes/no answer. Is it fair to assume that having T.38 setup properly, any fax machine will work? |
14:56.29 | *** part/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
14:57.25 | r0m|u | Should work* Yes. I would say with a 3% to 10% TX/RX failure. In a heavy used env. |
14:57.27 | irroot | FinboySlick yes that is the idea but not all devices work equally the SPA-2102 is a winner |
14:57.56 | FinboySlick | irroot: That's lucky, it's exactly what we planned to use. |
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14:58.49 | irroot | FinboySlick timing is critical so best use dahdi timing source the T38 gateway code was developed with testing on 2102 and HP-6500 multifunctional |
14:59.02 | FinboySlick | irroot: Just to be clear, T.38 essentially makes sure the analog portion of the fax transmission doesn't have to move beyond the ATA? |
15:00.28 | FinboySlick | apologizes, should really be reading that answer on wiki ;) |
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15:02.45 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-piaqnledfvzjzfln) |
15:02.45 | p3nguin | Hey, look! It's that Cisco guy, cbwest, again. |
15:04.29 | irroot | FinboySlick asterisk 1.4 allowed T.38 pass through so if your SIP provider allowed T.38 you could fax directly this was extended in 1.6 to allow T.38 termination of faxing via app_fax/res_fax |
15:05.21 | irroot | this has now been extended to allow "conversion" of audio fax "T30" on TDM lines ie PRI/BRI/FXO to T38 ie SPA-2102 |
15:05.46 | *** join/#asterisk krotos (~d3v1l@host211-3-dynamic.5-87-r.retail.telecomitalia.it) |
15:05.48 | FinboySlick | irroot: This setup will likely involve an SPA-2102 to a SwitchVox, to a MetaSwitch, to the world. |
15:05.53 | krotos | hi all guy |
15:05.57 | krotos | and girls |
15:07.06 | irroot | T30 faxes do not work on ethernet T38 allows this to work so where you need to have a fax machine on the network [T38] and the fax line is TDM this is now supported in asterisk-10 |
15:07.06 | FinboySlick | irroot: So basically, the SPA-2102 will convert things to T38 and it would ideally stay T.38 all the way to the MetaSwitch. |
15:07.44 | irroot | FinboySlick yes this has worked since 1.4 but better support in 1.6+ |
15:08.34 | FinboySlick | irroot: I should be safe with SwitchVox then... It's usually fairly up to date. |
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15:09.49 | *** mode/#asterisk [+o malcolmd] by ChanServ |
15:10.11 | bulkorok | Hi,... can I override allowguest=no for specific client-ips? |
15:11.51 | [TK]D-Fender | bulkorok, No, but yuo can make peers for them with "insecure=port,invite" |
15:12.47 | bulkorok | ok... thx |
15:12.59 | krotos | my voip server has two public ip, IP_1 and IP_2. Asterisk is bind on IP_1, but now i need to connect to a nortel pbx using IP_2 for sip (type peer). |
15:13.17 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
15:13.19 | krotos | how can i configure the peer , to use IP_2 for outbound traffic? |
15:13.48 | patrickod | I've tried setting canreinvite=no in general in sip.conf yet I'm still seeing Packet2Packet bridging |
15:15.47 | r0m|u | man Christmas is killing my pocket.... This new toys the kids want are just insane! "Beyblade" 10.00 dollars a pop! :/ lol |
15:16.06 | [TK]D-Fender | patrickod, What for of *? |
15:16.24 | bulkorok | [TK]D-Fender: insecure=invite is enough in my config :-) thx again! |
15:16.29 | patrickod | I'm trying to set up call forwarding by dialing outbound on a sip trunk |
15:16.38 | patrickod | the call is made outbound, it dials and picks up |
15:16.48 | patrickod | but there is no audio pased between either end |
15:16.51 | [TK]D-Fender | bulkorok, You're welcome |
15:17.04 | [TK]D-Fender | patrickod, What version of *? |
15:17.07 | [TK]D-Fender | (oops'd |
15:17.26 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
15:17.56 | patrickod | [TK]D-Fender: 1.6.2.9-2+squeeze3 |
15:18.02 | [TK]D-Fender | patrickod, you were already told to use "directmedia=no" for this. |
15:18.06 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
15:18.10 | [TK]D-Fender | patrickod, the parameter changed |
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15:20.34 | patrickod | [TK]D-Fender: I've set directmedia=no in general and still I'm seing Packet2Packet bridging |
15:21.07 | [TK]D-Fender | patrickod, pastebin your configs (all related portions) and the complete call attempt with SIP debug enabled |
15:21.11 | file | Packet2Packet bridging is not direct media |
15:21.31 | file | media still flows through Asterisk, it's just optimized internally |
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15:25.21 | patrickod | http://pastie.org/private/93wwhuzgrzok1lpgpxw7dg the log and config files |
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15:27.55 | patrickod | http://pastie.org/private/opmkpjtcm01xe1cvtd0rq last bit of the scrollback from sip debug |
15:28.03 | patrickod | includes the start of packet bridging |
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15:31.33 | patrickod | file: [TK]D-Fender any ideas why I'm not hearing audio ? |
15:32.24 | file | have you done a packet capture to confirm the flowing of audio? |
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15:33.19 | [TK]D-Fender | patrickod, <--- Reliably Transmitting (NAT) to 216.115.69.144:5060 ---> <-- flowroute is not behind NAT. Fix your peer |
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15:33.36 | patrickod | ok |
15:34.28 | patrickod | file: would sip show channelstats suffice ? |
15:34.41 | patrickod | it says nothing received or sent |
15:34.56 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
15:35.39 | patrickod | [TK]D-Fender: now I'm not hearing a dialtone ? |
15:36.04 | patrickod | and audio still isn't flowing |
15:38.02 | *** join/#asterisk ruied (~ruied@po-217-129-252-25.netvisao.pt) |
15:38.03 | *** join/#asterisk ddickenson (~ddickenso@67-198-0-5.static.grandenetworks.net) |
15:38.40 | [TK]D-Fender | patrickod, Validate each direction independently first |
15:38.49 | [TK]D-Fender | patrickod, As well as your forwarding. |
15:39.01 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
15:39.23 | [TK]D-Fender | patrickod, Then PB a new call with SIP debug. a complete call, not some half-way measure |
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15:39.53 | ddickenson | can someone point me to a person I can talk to offline or one on one about large system architecture and best practices, possibly also help me decide if I need to implement something like OpenSER for the size install I'm looking at? I've done tons of small installs but I'm looking at a 2500+ endpoint install for a Hospital that has to be up all the time and I'm a bit intimidated. |
15:41.18 | patrickod | [TK]D-Fender: Calls were worknig both inbound and outbound when I started |
15:41.27 | patrickod | it was only the bridging of the two that was not working. |
15:43.09 | ddickenson | anyone? |
15:45.14 | patrickod | http://privatepaste.com/8a6d30d275 full sip debug for the duration of a call |
15:46.42 | patrickod | I'm getting a 407 from flowroute ? |
15:47.56 | *** join/#asterisk X-Rob_ (~Rob@eth2083.qld.adsl.internode.on.net) |
15:48.34 | [TK]D-Fender | patrickod, Is that .... a question? |
15:49.06 | *** join/#asterisk serafie (~erin@nat/digium/x-wnmsxhnpbyayalfy) |
15:49.19 | patrickod | I'm confused as to why it would ask, i'm just executing a dial as per normal in the dialplan |
15:50.15 | [TK]D-Fender | Call out : Contact: <sip:+16507017829@66.201.49.164> |
15:50.31 | [TK]D-Fender | call in : INVITE sip:16505215946@50.18.181.199 SIP/2.0 |
15:50.51 | [TK]D-Fender | Not sure if I've mixed IP's here... but is your WAN IP right on your NAT settings? |
15:50.58 | [TK]D-Fender | And double check your port forwarding. |
15:51.03 | patrickod | yep |
15:51.05 | [TK]D-Fender | What do you have? |
15:51.21 | patrickod | I checked the WAN ip was correct in the NAT settings |
15:51.26 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
15:51.32 | patrickod | the IPs there are the 2 different trunking hosts |
15:51.44 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
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15:54.40 | *** part/#asterisk AmirBehzad (~behzad@31.184.187.2) |
16:00.39 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
16:00.41 | wcselby | o/ |
16:00.48 | [TK]D-Fender | <PROTECTED> |
16:02.00 | patrickod | [TK]D-Fender: to what extent could iptabels and FORWARD rules kill this ? |
16:02.36 | patrickod | I've noticed here that this box is -A FORWARD -j REJECT |
16:03.03 | p3nguin | It's a router? |
16:03.22 | [TK]D-Fender | patrickod, Nuke and find out |
16:04.28 | patrickod | yep it's port forwarding |
16:04.33 | patrickod | not on FORWARD but on INPUT |
16:04.47 | patrickod | now to figure out what ports need to be opened that wernet' already |
16:04.48 | wdoekes2 | NAT requires FORWARDing between the two nics |
16:04.58 | p3nguin | INPUT doesn't forward. |
16:05.05 | wdoekes2 | PREROUTING does |
16:05.08 | p3nguin | INPUT is for things destined for this system. |
16:05.19 | p3nguin | To route, you preroute and forward. |
16:05.19 | wdoekes2 | look at -t nat |
16:05.51 | p3nguin | iptables -t nat -L PREROUTING -nv |
16:05.58 | *** join/#asterisk irroot (~gregory@197.172.193.39) |
16:07.08 | patrickod | prerouting policy is accept |
16:07.39 | p3nguin | But if there are no rules, there is no port forwarding through the NAT. |
16:07.59 | patrickod | this is not a router |
16:08.24 | p3nguin | Why were you talking about port forwarding? |
16:09.15 | p3nguin | I just love to be given misinformation when being asked for help. It keeps me on my toes. |
16:11.12 | p3nguin | So... if it isn't a router (when you said it was, and was doing port forwarding), having a FORWARD policy of REJECT is pretty much a moot point. |
16:11.26 | *** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de) |
16:11.59 | p3nguin | If it isn't a NAT router, then the nat table and PREROUTING chains are not relevant. |
16:12.00 | patrickod | the box has iptables rules which seems to have been stopping calls having audio |
16:12.10 | patrickod | the INPUT chain is at fault it would seem |
16:12.24 | p3nguin | Yeah, INPUT and OUTPUT control what comes into and goes out of that host. |
16:12.25 | patrickod | giving a black -A INPUT -j ACCEPT has solved the silence problem |
16:13.03 | p3nguin | INPUT for things destined for that host, OUTPUT for things originating from that host |
16:13.33 | patrickod | I have the call live now and I'm trying to figure out which port its using |
16:13.36 | wdoekes2 | patrickod: look at rtp.conf and the ports there.. open those selectively with -p udp --dport start:end |
16:14.09 | patrickod | cool will do |
16:18.18 | patrickod | yep that's solved it |
16:18.41 | patrickod | I presume the rules let already established connectinos from these hosts inbound |
16:20.47 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
16:22.12 | p3nguin | If you have a rule for RELATED,ESTABLISHED it will. |
16:23.26 | irroot | need to load the state mod also good idea to do contrack |
16:23.35 | p3nguin | iptables -I INPUT -m state --state RELATED,ESTABLISHED -j ACCEPT |
16:24.56 | p3nguin | However, if your INPUT policy is ACCEPT, any rules that don't have a target of DROP or REJECT may never make a difference. |
16:24.57 | irroot | if you use contrac [RELATED] then the RTP will be allowed without any other special ruled with the sip contrack helper |
16:26.02 | voipeng | any way to isolate why when I go to record an AA greeting using *321 it establishes the call, but it doesnt allow me to record anything? |
16:26.27 | p3nguin | core set verbose 3 |
16:26.35 | p3nguin | Make the call. Show us what happened. |
16:26.39 | voipeng | k |
16:26.53 | *** join/#asterisk timahvo1 (~rogue@197.176.25.176) |
16:28.47 | voipeng | http://pastebin.com/hPnns28E |
16:29.54 | leifmadsen | permissions issue in folder potentially |
16:29.59 | leifmadsen | can't create the file |
16:30.09 | leifmadsen | or the directory doesn't exist you're trying to create the file in |
16:30.29 | leifmadsen | <PROTECTED> |
16:30.38 | [TK]D-Fender | IIRC It should create a file in the full path if there are permissions up to the point where the path doesn't exist |
16:30.57 | leifmadsen | ^^^ |
16:31.01 | voipeng | leifmadsen: thanks, so i guess i should manually create the directories? |
16:31.06 | leifmadsen | obviously |
16:31.10 | voipeng | the problem is im tryin to record |
16:31.12 | voipeng | and make it |
16:31.17 | voipeng | not play an existing file |
16:31.20 | leifmadsen | that's not asterisk's problem to resolve |
16:31.23 | voipeng | heh |
16:31.25 | [TK]D-Fender | go prove what user owns the foders through that path |
16:31.25 | voipeng | thanks |
16:31.45 | leifmadsen | use STAT() to check if the directory exists, and if not, then use SHELL(mkdir -p /my/path/) |
16:31.46 | voipeng | k |
16:33.33 | voipeng | yea it didnt exist, i guess ill try and work with voiceaxis more to see why its not creating, not sure i want to get into making directories each time someone goes to make a new greeting |
16:34.05 | leifmadsen | ya that is a sysadmin issue -- not an asterisk issue |
16:34.13 | voipeng | k |
16:34.16 | voipeng | thank you |
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16:42.23 | *** join/#asterisk jcook_5xdata (~jcook_5xd@173.162.32.1) |
16:42.56 | jcook_5xdata | is there a way to stop dynamic creation of meetme rooms |
16:43.44 | p3nguin | Yes. Don't use the option that makes dynamic conferences. |
16:45.16 | [TK]D-Fender | Doctor, Doctor! It hurts when I rai.... erm ... nevermind ... |
16:45.45 | jcook_5xdata | I am not sure what you mean. I have no reference to meetme in any dial plans, but somehow ghost are creating a room with two people in there |
16:46.14 | jcook_5xdata | is there a option in meetme.conf dynamic= no |
16:46.48 | [TK]D-Fender | jcook_5xdata, Can you show us the problem? |
16:48.17 | jcook_5xdata | be a sec I kick the room. I have to wait till it created again |
16:48.30 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
16:48.53 | p3nguin | While you're waiting for that, pastebin your "dialplan show" output. |
16:49.12 | voipeng | i know we established my AA recording was a voiceaxis issue, id like to know if there app_record had permissions I can view or eyeball against a working server... |
16:49.59 | [TK]D-Fender | voipeng, it isn't "record".. its the * user |
16:50.29 | [TK]D-Fender | voipeng, You don't compare to others.. you compare to yourself |
16:50.33 | voipeng | hah |
16:50.37 | *** join/#asterisk mattchis (~mattchis@75-145-122-77-Colorado.hfc.comcastbusiness.net) |
16:50.40 | p3nguin | "ps -C asterisk u" |
16:50.46 | voipeng | i have other pbx that work |
16:51.08 | [TK]D-Fender | voipeng, Again, they might nt be running as the same user. You should not be comparing these |
16:51.14 | voipeng | k |
16:51.16 | p3nguin | "namei -m /var/lib/asterisk/sounds/aa/lctcap/" |
16:51.33 | *** join/#asterisk LemensTS (~matthew@70.238.163.254) |
16:51.50 | *** part/#asterisk mattchis (~mattchis@75-145-122-77-Colorado.hfc.comcastbusiness.net) |
16:52.29 | LemensTS | If a telco brings a t1 line into a business, do they just run it into a patch panel and send you a t1 router? What I was wondering is how to run the wire from the patch panel to the t1 router. Or does the patch panel usually have a rj45 port? |
16:52.36 | voipeng | namei -m /var/lib/asterisk/sounds/aa/lctcap |
16:52.37 | voipeng | f: /var/lib/asterisk/sounds/aa/lctcap |
16:52.37 | voipeng | <PROTECTED> |
16:52.37 | voipeng | <PROTECTED> |
16:52.37 | voipeng | <PROTECTED> |
16:52.38 | voipeng | <PROTECTED> |
16:52.40 | voipeng | <PROTECTED> |
16:52.42 | voipeng | <PROTECTED> |
16:52.44 | voipeng | <PROTECTED> |
16:52.44 | jcook_5xdata | p3nguin, here it is http://pastebin.com/PARB4qbH it is very simple |
16:52.58 | p3nguin | Hey, [tk]d-fender, my car runs fine... I change the oil regularly, run premium fuel, and even wash it when it isn't raining. Now, can you tell me what's wrong with my truck? |
16:53.37 | p3nguin | voipeng: Dammit, I did it again. I meant namei -mo /var/lib/asterisk/sounds/aa/lctcap/ |
16:54.15 | voipeng | <PROTECTED> |
16:54.16 | voipeng | namei: invalid option -- o |
16:54.16 | voipeng | usage: namei [-mx] pathname [pathname ...] |
16:54.16 | p3nguin | This is the second time this month I have done that. |
16:54.24 | [TK]D-Fender | voipeng, PASTEBIN <- |
16:54.25 | [TK]D-Fender | ~pb |
16:54.26 | 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. |
16:54.28 | *** join/#asterisk r0m|u (~wtf@darkstar.rice.edu) |
16:54.34 | voipeng | sorry! didnt think 5 lines was alot |
16:54.47 | p3nguin | It isn't... it's "a lot" |
16:54.47 | [TK]D-Fender | 9 <- |
16:54.53 | voipeng | hahah |
16:54.56 | *** join/#asterisk celord (~celord@201.195.243.194) |
16:54.57 | voipeng | k |
16:55.16 | p3nguin | hmm, well, namei should be able to show the modes and the owners. I don't know what kind of namei you have that doesn't work right. |
16:55.40 | p3nguin | http://pastebin.com/Dx1gmhPM |
16:56.03 | voipeng | hmm |
16:56.12 | voipeng | yea that is weird, i did it as sudo as well same output |
16:56.34 | p3nguin | Running it as a different user isn't going to change the options it has. |
16:57.14 | p3nguin | I'd like to see -mo or -l to see the modes and owners in one go. Alternatively, you can use ls -dl on each directory down the path. |
16:59.19 | voipeng | well im pretty sure the files arent there, so there wouldnt be an owner |
16:59.19 | *** join/#asterisk r0m|u (~wtf@darkstar.rice.edu) |
16:59.47 | p3nguin | Why aren't you creating any dirs that are required but not present? |
17:00.01 | voipeng | http://pastebin.com/hPnns28E |
17:00.04 | voipeng | it tries to |
17:00.04 | LemensTS | Do you just use a normal ethernet cable from T1 jack from telco to cisco router? |
17:00.11 | voipeng | but i get app_record.c:272 record_exec: Could not create file |
17:00.28 | LemensTS | nm found a cisco room |
17:00.40 | *** part/#asterisk LemensTS (~matthew@70.238.163.254) |
17:00.43 | voipeng | rollover/flat cable |
17:00.44 | voipeng | heh |
17:00.49 | p3nguin | app_record.c:272 record_exec: Could not create file /var/lib/asterisk/sounds/aa/lctcap/1_temp <----- |
17:00.58 | voipeng | yea.. |
17:01.08 | p3nguin | It can't create the file if ANY OF THE DIRECTORIES is not present. |
17:01.13 | voipeng | oh |
17:01.25 | p3nguin | lctcap is not there. Why haven't you created it yet? |
17:01.29 | voipeng | guess i totally misinterrprettted that |
17:01.44 | voipeng | we never have to manually create these entries |
17:01.50 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
17:02.00 | p3nguin | Great, so your system works and there is no more question. Enjoy! |
17:02.11 | voipeng | that would be an administrative nightmare if everytime someone went to record we would need to make a directory |
17:02.12 | voipeng | lol |
17:02.24 | r0m|u | what a day |
17:02.28 | voipeng | mmhmm |
17:03.28 | p3nguin | If the directory can be created automatically, it will be done as the user which asterisk runs as. If the parent directory is not writable by asterisk user, it won't work. |
17:03.40 | p3nguin | hence my namei -mo request. |
17:04.39 | r0m|u | waz up p3nguin |
17:04.44 | p3nguin | Since your namei sucks, check the ownership of the directories to make sure the user can write there. To see which user runs asterisk, check "ps -C asterisk u". |
17:05.05 | p3nguin | (should be "asterisk" in most cases) |
17:06.24 | jcook_5xdata | I dont know maybe when I kick the room the ghost got scared and found anything box to live in |
17:08.13 | [TK]D-Fender | <voipeng> that would be an administrative nightmare if everytime someone went to record we would need to make a directory <- that's funny.. because your custom dialplan explicitly names that directory. I |
17:09.11 | voipeng | hmm? where |
17:10.20 | leifmadsen | we're still talking about this? |
17:10.35 | voipeng | im talking to the software provider now |
17:10.41 | voipeng | sorry |
17:10.53 | leifmadsen | don't be sorry |
17:11.05 | leifmadsen | I just thought it was already resolved as far as asterisk was concerned |
17:12.04 | voipeng | it is, i was looking for any additional information to give to the software provider in the interim |
17:12.13 | voipeng | i can certainly post what was the issue once its resolved |
17:13.16 | *** part/#asterisk patrickod (~Patrick@lysander.patrickodoherty.com) |
17:13.56 | p3nguin | r0m|u: I got my lab results... Sorry, but I'm not going to be able to donate my lungs this month. :) Results: total cholesterol 167, ldl cholesterol 107.4, triglycerides 83, alt 52 (liver), ast 18 (liver), creat 1.5 (kidney). |
17:14.25 | p3nguin | Prognosis: I'M GOING TO LIVE! |
17:14.48 | carrar | Better start hiking every day |
17:15.02 | p3nguin | Kidneys are still not too good, but they aren't any worse than they were a year ago. |
17:15.13 | p3nguin | Hiking? Why? |
17:15.25 | carrar | You need good excersize |
17:15.28 | carrar | You need good excersize program |
17:16.15 | p3nguin | I know i need exercise, but my numbers are all good, with the exception of the kidney being a bit out of whack. |
17:16.50 | carrar | "I don't need to wear a seat belt! I haven't been in a accident yet" |
17:17.26 | p3nguin | I'm not saying I can just let myself go because I have good results. |
17:17.28 | p3nguin | Not saying that at all. |
17:17.41 | carrar | Oh I think you ARE saying that |
17:17.46 | p3nguin | But you came off as I need to ramp up my routine because my numbers are bad. |
17:17.47 | carrar | I heard it LOUD AND CLEAR!! :) |
17:18.01 | carrar | Oh yeah |
17:18.03 | carrar | You do |
17:18.09 | p3nguin | But the numbers are good. |
17:18.14 | carrar | You need to start walking 40 miles aday |
17:18.31 | p3nguin | I don't even have time to walk 40 miles! |
17:18.36 | [TK]D-Fender | AND I WOULD WALK 500 MORE! |
17:18.44 | carrar | Back in my great great great granpa's day they used to walk 40 miles up hill through snow to get to work! |
17:18.59 | [TK]D-Fender | messes with peoples musical sub-conscious ... |
17:19.11 | carrar | heh |
17:19.15 | p3nguin | up hill BOTH ways. |
17:19.19 | carrar | YEAH! |
17:19.28 | [TK]D-Fender | in show up to our eyeballs! |
17:19.56 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
17:19.58 | p3nguin | So, anyway, I'm going to live for a while longer, |
17:20.15 | carrar | You've live to 29? |
17:20.16 | [TK]D-Fender | Kiddo you don't how good ja got it... in my day, before electricity we had to watch television by CANDLELIGHT |
17:20.19 | carrar | (Logans Run) |
17:20.59 | p3nguin | I did live to 29. But I don't remember ever watching that movie. :/ |
17:21.49 | chuckf | I know I watched the movie but have blocked most of it out |
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17:28.34 | anonymouz666 | is there a command that cleans the whole astdb? |
17:28.41 | anonymouz666 | database deltree all |
17:28.44 | anonymouz666 | something like that |
17:28.52 | anonymouz666 | or "rm astdb" |
17:28.55 | p3nguin | rm /var/lib/asterisk/astdb |
17:29.02 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-qhvvcnzevmmdbfey) |
17:29.15 | p3nguin | Then restart asterisk so it gets recreated fresh. |
17:29.38 | anonymouz666 | alright, that was what I thought |
17:36.30 | r0m|u | p3nguin: !!!!!!!!!!!! awesome news man. |
17:37.11 | r0m|u | r0m|u: test |
17:37.37 | r0m|u | thats very good news p3nguin |
17:39.13 | p3nguin | Yep, he's going to let me live. |
17:39.21 | r0m|u | :) |
17:40.25 | *** join/#asterisk bolkin (~bmint@h174.92.190.173.static.ip.windstream.net) |
17:45.08 | r0m|u | r0m|u: test |
17:45.53 | *** join/#asterisk Steel_Reign (~steel@207.239.162.198) |
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17:50.27 | r0m|u | r0m|u: test |
17:54.04 | r0m|u | r0m|u: test |
17:54.40 | *** join/#asterisk r0m|u (~wtf@darkstar.rice.edu) |
17:58.23 | *** join/#asterisk r0m|u (~wtf@darkstar.rice.edu) |
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17:59.01 | r0m|u | r0m|u: test |
18:00.07 | WIMPy | r0m|u: Have you created an endless loop, replying to your own test? |
18:00.21 | r0m|u | WIMPy: lol |
18:00.52 | r0m|u | I just finish setting up "notification" :) |
18:01.12 | r0m|u | WIMPy: And thank you you prove that it worked |
18:01.43 | WIMPy | If only everything was that easy. |
18:02.09 | r0m|u | :) |
18:09.19 | *** join/#asterisk godmachine-x6 (~poorelanc@funtoo/user/godmachine-x6) |
18:16.02 | wcselby | anyone know of any alternatives to navicat that are free / open source that will work with older versions of mysql? |
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18:23.39 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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18:24.22 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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18:28.57 | r0m|u | p3nguin: I got a reply from freenum.... They did a large import and by mistake they left my account out. In top of that the admin by accidnet overlooked my account and did not approved it causing the issues that we saw yesterday. He said is fixed and should be in their dns shortly. |
18:29.04 | MrTelephone | how do you get directory() working with odbc voicemail configurations? |
18:29.48 | *** join/#asterisk PoWeRKiLL (~powerkill@77.125.87.36) |
18:30.07 | jcook_5xdata | p3nguin, hey ghost are back. remember I had the ghost creating meetme rooms |
18:30.11 | p3nguin | Over 1500 registrants, and he left yours out. You believed that? |
18:30.26 | p3nguin | jcook_5xdata: core show channels verbose |
18:30.37 | p3nguin | meetme list |
18:30.43 | MrTelephone | When I switched to the voicemail_users database I cannot dial by first name anymore :( |
18:30.56 | p3nguin | And I'm still waiting on the output from "dialplan show" |
18:31.46 | r0m|u | p3nguin: LOL |
18:32.39 | jcook_5xdata | here is the meetme list http://pastebin.com/prJetrjG |
18:33.53 | [TK]D-Fender | And a channel list |
18:33.58 | [TK]D-Fender | "core show channels" |
18:34.02 | jcook_5xdata | they are gone again so I could not get channels... waiting |
18:34.13 | [TK]D-Fender | get the rest |
18:34.21 | MrTelephone | My database merge didn't work worth the shit |
18:34.23 | MrTelephone | fullname is missing |
18:35.38 | jcook_5xdata | here is the core show http://pastebin.com/HvfgbDv7 |
18:36.16 | [TK]D-Fender | no, no meetme there so far |
18:36.23 | [TK]D-Fender | Waiting on dialplan... |
18:37.02 | jcook_5xdata | from me? |
18:43.59 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
18:44.27 | jcook_5xdata | here is my full extension.conf http://pastebin.com/b6CLCcYx |
18:44.46 | *** join/#asterisk PoWeRKiLL (~powerkill@77.125.87.36) |
18:44.59 | MrTelephone | hmm dial by name is still not working. Is odbc supported for dial by name in asteirsk 1.8.5? |
18:46.26 | [TK]D-Fender | jcook_5xdata, "core show application meetme" <- |
18:47.50 | gordonjcp | to paraphrase the old old internet meme, I've lost a phone somewhere in my house, literally lost it |
18:48.00 | gordonjcp | I can ring it but I can't work out where the noise is coming from |
18:48.30 | gordonjcp | oh, okay, found it, now this is good |
18:48.46 | gordonjcp | I've packed it away in a box with some other stuff - while it was still plugged into the fxs |
18:48.49 | jcook_5xdata | [TK]D-Fender, just show a man page |
18:48.52 | gordonjcp | o_O |
18:49.04 | [TK]D-Fender | PB <- |
18:50.13 | Steel_Reign | question sorry for the noobness but i am new to asterisk. I have already got a server up and running. my question is that how many calls can i make on one sip line? |
18:50.32 | [TK]D-Fender | no such thing as a "sip line" |
18:50.41 | Steel_Reign | ok on one sip then |
18:51.09 | [TK]D-Fender | You send calls over SIP. Typically with auth info tied to an account. How many calls you are allowed over that account depends on what service you are being offered |
18:51.15 | [TK]D-Fender | And we don't know |
18:51.31 | gordonjcp | Steel_Reign: how many web pages can you open with one internet connection? |
18:51.44 | [TK]D-Fender | It's probably spelled out pretty clearly in the terms of service for whatever you're paying for. |
18:52.01 | Steel_Reign | i have not got one yet |
18:52.18 | [TK]D-Fender | well it depends on what service you pick |
18:52.33 | [TK]D-Fender | How many slices are there in a pre-sliced loaf of bread? Depends. |
18:52.54 | Steel_Reign | so then i have have 1 sip account and have a office of 50 people make outbound calls on that one sip? |
18:53.35 | [TK]D-Fender | You COULD... |
18:53.45 | [TK]D-Fender | again depends on what kind of service you pay for |
18:53.52 | Steel_Reign | understood |
18:54.17 | [TK]D-Fender | Could be 1, could be 50. Could be "abitrarily large up to carrier and chain capacity" |
18:55.54 | [TK]D-Fender | Steel_Reign, typically you pay for X # of simultaneous channels (calls), X # of unique inbound phone numbers (DID) to process calls in the ways you want. Might only be 1 DID with a provider tha lets you call out 50 times |
18:56.47 | [TK]D-Fender | Some providers only offer products that resemble analog line capabilities. 2 calls at a time, 1 DID for that link and account, and if you want more, then they come in as different phone #'s |
18:57.33 | Steel_Reign | "call out 50 times" does that mean simultaneous connections of 50 individual calls out? |
18:58.02 | jcook_5xdata | [TK]D-Fender, ok here is the core show Channel http://pastebin.com/XmPzKZND You know what I think it is paging intercom? maybe.. |
18:59.15 | p3nguin | Page() probably uses MeetMe, then. |
18:59.41 | [TK]D-Fender | Steel_Reign, 50 simultaneous calls... any combination of in/out really |
18:59.54 | [TK]D-Fender | Correct |
19:00.51 | Steel_Reign | ok. is a sip required to make calls to other asterisk servers around the world? |
19:02.58 | [TK]D-Fender | Steel_Reign, Ok, lets fix up some basics : SIP is a protocol. Not a "thing". Like my sending an e-mail. that is a protocol, but not a specific service or piece of software. |
19:03.32 | Steel_Reign | k |
19:03.34 | [TK]D-Fender | Steel_Reign, SIP is a protocol for placing calls (voice/video, etc) |
19:04.05 | [TK]D-Fender | Steel_Reign, there are hardware phones that use SIP, and of course * uses it as well. This means that you can certainly call direct from one * box to another. |
19:04.06 | Steel_Reign | got it |
19:05.00 | [TK]D-Fender | There are other protocols : IAX2 (Inter-Asterisk eXchange), H.323 that can also be used, but SIP is the most popular in general, and IAX used in many cases between 2 * boxes |
19:05.14 | [TK]D-Fender | These are all VoIP protocols. |
19:05.41 | [TK]D-Fender | * can also use special hardware to talk to a variety of physical telephony lines. |
19:07.42 | Steel_Reign | ok so a basic asterisk install of asterisk on a server by itself can call any voip system provided that its configure correctly with no third party service providers right? |
19:07.50 | Steel_Reign | call from box to box |
19:08.22 | MrTelephone | I wonder why there is no app_directory in the asterisk source folder? |
19:09.13 | Steel_Reign | sorry for all the ?'s. like i said i am new to this. picked it up little more then a week ago. |
19:10.24 | [TK]D-Fender | Steel_Reign, Correct. SIP is SIP. Any 2 systems or devices can pretty much talk to each toher as long as configured on each side |
19:11.48 | [TK]D-Fender | Steel_Reign, And VoIP providers are just 1 way of reaching the PSTN (real world phone system) |
19:11.56 | voipeng | any suggestions why a number would give me an error: sent into invalid extension 's' |
19:12.24 | [TK]D-Fender | Steel_Reign, What vendor and technology is the best choice for you will depend on a number of factors. |
19:12.39 | [TK]D-Fender | voipeng, It's looking for "s". It doesn't exist, just like it says |
19:13.05 | voipeng | d-fender: so i need to see if its defined in extensions.conf ? |
19:15.04 | Steel_Reign | thanks D-Fender |
19:15.40 | jcook_5xdata | voipeng, yes look at the error it will state what number or ext it looking for then do something like exten => number it looking for,1,answer() then exen = |
19:15.55 | voipeng | thanks |
19:16.00 | jcook_5xdata | > #,2, whatever |
19:16.41 | MrTelephone | qwell, you around? |
19:19.58 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
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19:25.27 | p3nguin | If it is looking for s, it's probably configured incorrectly in the first place. |
19:25.46 | p3nguin | Not to mention extension s isn't there, which is pretty normal. |
19:26.54 | p3nguin | What's the name of the song by Cross Canadian Ragweed where he sings "I got sober, now it's over, I'm back to drinkin' again"? |
19:27.21 | p3nguin | or rather "it's back to drinkin' again" |
19:28.30 | [TK]D-Fender | <voipeng> d-fender: so i need to see if its defined in extensions.conf ? <- it clearly isn't. That's the point. |
19:28.52 | voipeng | ? |
19:28.58 | voipeng | i fixed it i did an extensions reload from the cli |
19:29.02 | p3nguin | error: sent into invalid extension 's' <--- it IS NOT defined. |
19:29.05 | voipeng | so yea |
19:29.10 | voipeng | thanks |
19:29.27 | MrTelephone | I had hidefromdir=yes by default :( |
19:32.02 | gordonjcp | p3nguin: a quick google suggests it's called "Drinkin' Song" |
19:32.21 | p3nguin | It didn't seem very definitive, though. |
19:40.41 | *** join/#asterisk madsage (~sage@io.ioio.com) |
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19:51.38 | r0m|u | p3nguin: It's answering now. |
19:52.02 | p3nguin | What is? |
19:52.17 | r0m|u | my frenum dns |
19:52.29 | r0m|u | s/frenum/freenum/ |
19:52.39 | *** part/#asterisk madsage (~sage@io.ioio.com) |
19:53.27 | p3nguin | Yep, I see the DNS is updated now. |
19:53.38 | r0m|u | cool. :) |
19:53.45 | p3nguin | And it is propagated to me, even. |
19:53.53 | r0m|u | can you try and dial it? |
19:53.56 | r0m|u | sweet! |
19:54.01 | p3nguin | 2231003? |
19:54.51 | p3nguin | I guess that's it. I'll try it. |
19:55.02 | r0m|u | Yes |
19:55.19 | p3nguin | It's a bunch of numbers to dial, I know that much. |
19:55.29 | p3nguin | 0122231003*xxxx |
19:56.02 | r0m|u | you should get vmail..... |
19:56.09 | r0m|u | that is. |
19:56.13 | p3nguin | Yep. Left a message. |
19:56.24 | r0m|u | cool. |
19:56.38 | r0m|u | well I think I am going to dial it down to the two digits |
19:56.42 | r0m|u | that is long as shit |
19:57.20 | r0m|u | as soon as my brother allowsme ill put in the company info instead of my info.... |
19:57.28 | r0m|u | s/allowsme/allows me/ |
20:01.07 | p3nguin | On the SPA 3102, which setting turns up the heard volume of the handset's mic? Calls from the phone are super quiet and hard to hear the person on it. |
20:10.18 | *** join/#asterisk irroot (~gregory@197.105.107.161) |
20:12.33 | *** join/#asterisk n3hxs (~ed@63.68.135.4) |
20:16.28 | r0m|u | p3nguin: Port Output Gain? |
20:17.29 | MrTelephone | FXS_Port_Input_Gain "-3" ; |
20:17.29 | MrTelephone | FXS_Port_Output_Gain |
20:17.42 | MrTelephone | too bad it doesn't have a setting for each port |
20:17.51 | MrTelephone | that's from a pap2t config though |
20:18.34 | r0m|u | p3nguin: Port output and input gain are under Regional. |
20:19.07 | r0m|u | "Miscellaneous" |
20:20.17 | r0m|u | p3nguin: here are som tips for the PAP2 which should also work with the SPA..... Since they are regional they are pretty much standards http://www.freepbx.org/support/documentation/howtos/how-to-set-up-a-linksys-pap2-or-sipura-spa-2000-for-use-with-freepbx |
20:21.22 | *** join/#asterisk cbwest (~cbwest@nat/cisco/x-pqefvpvtpbjogkla) |
20:21.28 | p3nguin | Hey, look! It's that Cisco guy, cbwest, again. |
20:25.33 | *** join/#asterisk As001 (~uros@cable-89-216-191-22.dynamic.sbb.rs) |
20:26.44 | r0m|u | As001: how is your hunt for your new system going? |
20:32.28 | MrTelephone | what mr poppers penguin? |
20:37.52 | r0m|u | pupers |
20:40.23 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
20:40.26 | wcselby | :( |
20:40.34 | wcselby | I just spilled my entire coke on the floor |
20:40.53 | wcselby | it was a good 38 oz left in the cup too |
20:42.17 | *** join/#asterisk cerberus_za (~coert@8ta-151-32-212.telkomadsl.co.za) |
20:43.11 | drudge` | holy crap |
20:43.19 | drudge` | no drinks or food in the noc, wcselby |
20:43.27 | wcselby | lol |
20:43.37 | wcselby | i'm in my office, but the office happens to be in the owner's house |
20:44.09 | wcselby | it was my fault for trying to work through lunch, nothing ever good comes from that |
20:47.00 | MrTelephone | wipe the mustard off your shirt bro |
20:48.20 | *** join/#asterisk cerberus_za (~coert@8ta-151-32-212.telkomadsl.co.za) |
20:50.42 | *** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
20:50.42 | *** join/#asterisk PoWeRKiLL (~powerkill@77.125.87.36) |
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20:51.18 | iprouteth0 | to record custom IVR annoucements in g.729, does it require a codec license? |
20:52.58 | [TK]D-Fender | no |
20:53.08 | [TK]D-Fender | Not if that's the codec of the channel recording |
20:53.47 | *** join/#asterisk timahvo1 (~rogue@197.177.128.196) |
20:57.14 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-oojhtbsuikhgditt) |
20:57.28 | *** join/#asterisk serafie (~erin@nat/digium/x-cofwfnipicfbqbge) |
20:59.51 | p3nguin | Record() won't have to translate it through slin? |
21:00.10 | p3nguin | I figured it would, like several other apps do. |
21:01.31 | [TK]D-Fender | Shouldn't |
21:01.34 | MrTelephone | my dialplan is not even funny anymore |
21:01.49 | [TK]D-Fender | You choose G.729 and are G.729 it should dump the packets straight |
21:01.58 | [TK]D-Fender | MrTelephone, It was funny before? |
21:02.00 | As001 | <PROTECTED> |
21:02.05 | MrTelephone | no |
21:02.29 | MrTelephone | i pushed everything into macros and it cleaned a lot of stuff up |
21:03.19 | *** part/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
21:04.45 | r0m|u | wcselby: screw the owners floor! The coke! You spilled it! sweet nectar of life! |
21:05.07 | wcselby | r0m|u tell me about it |
21:05.18 | r0m|u | lol |
21:05.39 | wcselby | i wasn't happy |
21:05.42 | wcselby | heh |
21:05.43 | r0m|u | coke is my every day coffe.... lol we all have owr poisons :P |
21:06.11 | wcselby | it's funny, because I never eat here in the "office", because it feels weird eating in this dude's house by myself |
21:06.23 | r0m|u | That sucks :( Been there though.... |
21:06.26 | wcselby | but today I was working on something and was in the groove so I didn't want to put an hour pause into it |
21:06.37 | r0m|u | And the day you do..... tan tan taaaan! |
21:06.40 | wcselby | which is what I ended up pretty much doing anyways |
21:07.18 | r0m|u | :( |
21:07.40 | wcselby | ah well, that'll learn me |
21:07.41 | wcselby | :) |
21:07.53 | r0m|u | :P |
21:08.53 | *** part/#asterisk Steel_Reign (~steel@207.239.162.198) |
21:09.28 | p3nguin | My FXS port input gain is 0. Shall I turn it up to 3? |
21:10.06 | p3nguin | The FXS port output gain is -6. |
21:10.46 | r0m|u | p3nguin: Yes |
21:10.59 | p3nguin | I would think input means from the phone's mic, and output would be to the phone's speaker. |
21:11.06 | p3nguin | I should turn them both up. |
21:11.11 | r0m|u | p3nguin: mine is -3 |
21:11.28 | p3nguin | I'm going to go to 3 and -3. |
21:11.30 | p3nguin | instead of 0 and -6. |
21:11.38 | r0m|u | cool |
21:12.40 | p3nguin | We'll see if that helps. |
21:13.07 | r0m|u | cool |
21:19.15 | *** join/#asterisk s[X] (~mark@ppp118-208-103-144.lns20.bne4.internode.on.net) |
21:21.40 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
21:28.22 | leifmadsen | p3nguin: I suggest turned everything to 11 |
21:29.40 | r0m|u | leifmadsen: That high? |
21:29.46 | r0m|u | is it a us standard? |
21:29.50 | leifmadsen | o.O |
21:30.00 | r0m|u | lol |
21:30.34 | leifmadsen | someone hasn't seen This Is Spinal Tap |
21:32.12 | *** part/#asterisk As001 (~uros@cable-89-216-191-22.dynamic.sbb.rs) |
21:33.00 | r0m|u | I am curious as of what settings are Optimal for the "Regional Settings" for the US. I had to change quite a few settings to match the US |
21:35.07 | [TK]D-Fender | leifmadsen, amp capo get yours now... |
21:35.18 | [TK]D-Fender | checkout time, BBIAB |
21:35.24 | leifmadsen | r0m|u: 11 probably isn't it |
21:35.41 | leifmadsen | and I'm sure the value heavily depends on what network you're on, and where you're located |
21:36.26 | r0m|u | leifmadsen: I figure after a quick google of "This Is Spinal Tap" |
21:36.31 | r0m|u | lol |
21:40.15 | *** join/#asterisk miztic (~miztic@75-149-203-105-Illinois.hfc.comcastbusiness.net) |
21:40.38 | r0m|u | just out of curiosity does most of you guys deny msgs? (Not a trick question) |
21:41.12 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
21:46.18 | Qwell | r0m|u: I read them, and then I potentially laugh at the person. |
21:46.28 | Qwell | A lot of people don't like unsolicited messages. |
21:46.36 | r0m|u | lol |
21:46.41 | r0m|u | I am sure |
21:47.04 | r0m|u | I was asking because as of lately I been getting a lot of msgs and I can understand how it can be annoying quick |
21:47.20 | Qwell | Shame them publicly. |
21:47.48 | r0m|u | nice. I shall learn from that masters! |
21:47.54 | r0m|u | bows |
21:47.58 | r0m|u | lol :P |
21:48.17 | leifmadsen | after getting the message, I close it, then come to the channel they messaged me from (usually this one), and say, "SomeCrazyHandle: don't message me directly -- ask publicly so other people can also help you and so that others may learn if I choose to respond" |
21:48.48 | Qwell | leifmadsen: exactly |
21:48.55 | Qwell | ~msg |
21:48.55 | infobot | (1) Use private messages to the bots to reduce channel spam, but don't message people on #debian without asking permission first. Most questions should be asked on channel, so that others can benefit from the question and the answers received. (2) Always feel free to message freenode network staff. They're the people with hostnames ending in 'staff.freenode'. (3) Monosodium glutamate, a food additive (see http://truthinlabeling.org/). |
21:49.25 | r0m|u | lmao nice |
21:49.31 | r0m|u | #3 |
21:49.38 | Qwell | that's nothing |
21:49.40 | Qwell | ~qwell |
21:49.40 | infobot | you are probably a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. qwellcommunicationsinc, the holding company of telcomjoshleifvoxmartinc |
21:50.02 | r0m|u | BAUHAHAHA! LOL! |
21:50.07 | r0m|u | ROFL |
21:50.13 | tzanger | what the ... |
21:50.28 | tzanger | I'm a little upset there's no zangmegacorp in that name |
21:50.28 | r0m|u | I cant stop laughing'1! |
21:50.30 | leifmadsen | telcomjoshleifvoxmartinc: you buy now! |
21:50.59 | p3nguin | ~leif |
21:51.03 | p3nguin | ~leifmadsen |
21:51.03 | infobot | i guess leifmadsen is blitzrage |
21:51.13 | p3nguin | Ya don't say... |
21:51.18 | r0m|u | LOL |
21:51.20 | leifmadsen | ~blitzrage |
21:51.20 | infobot | [blitzrage] a super cool fellow |
21:51.24 | pabelanger | infobot: leifmadsen+ |
21:51.31 | Qwell | ~karma leifmadsen |
21:51.31 | infobot | leifmadsen has karma of -1 |
21:51.33 | Qwell | ~karma qwell |
21:51.33 | infobot | qwell has karma of 11 |
21:51.35 | leifmadsen | infobot: pabelanger ++ |
21:51.38 | Qwell | Get. Owned. |
21:51.39 | leifmadsen | infobot: pabelanger++ |
21:51.47 | pabelanger | infobot: Qwell-- |
21:51.57 | r0m|u | hahaha I had no clue of this.... |
21:51.59 | leifmadsen | ~karma pabelanger |
21:51.59 | infobot | pabelanger has karma of 4 |
21:52.04 | pabelanger | infobot: tell Qwell his is lame |
21:52.05 | leifmadsen | wtf -1?! |
21:52.11 | Qwell | ~karma blitzrage |
21:52.11 | infobot | blitzrage has karma of 1 |
21:52.13 | r0m|u | lol! |
21:52.16 | tzanger | infobot: leifmadsen -- |
21:52.22 | leifmadsen | tzanger: failsauce |
21:52.29 | tzanger | apparently |
21:52.29 | p3nguin | -1 and 1 kind of average out to 0. |
21:52.35 | r0m|u | haha |
21:52.42 | tzanger | messages the bot to get leif down to a -50 |
21:52.46 | pabelanger | tell me about leifmadsen |
21:52.55 | pabelanger | tell me about infobot |
21:53.01 | p3nguin | Is it nap time yet? |
21:53.05 | leifmadsen | divide by zero |
21:53.49 | p3nguin | I have drugs; maybe drugs will make it nap time. |
21:53.58 | pabelanger | infobot: hex carrot |
21:53.58 | infobot | carrot is 63 61 72 72 6F 74 |
21:54.00 | r0m|u | lmao!!! |
21:54.22 | p3nguin | More Flintstone chewable Morphine, please! |
21:54.25 | pabelanger | infobot: karma Qwell |
21:54.25 | infobot | qwell has karma of 10 |
21:54.59 | r0m|u | ~wtf |
21:55.00 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
21:55.07 | *** join/#asterisk s[X] (~mark@eth589.qld.adsl.internode.on.net) |
21:55.19 | pabelanger | infobot: wtf iirc |
21:55.21 | r0m|u | Desc: Interface to the BSD wtf command |
21:56.28 | *** join/#asterisk jkroon (~jkroon@dsl-241-250-126.telkomadsl.co.za) |
21:56.28 | r0m|u | all right.... Time to go home! cya in a bit! |
21:56.30 | *** part/#asterisk FinboySlick (~shark@74.117.40.10) |
21:57.24 | Qwell | Qwell++ |
21:57.26 | Qwell | ~karma qwell |
21:57.26 | infobot | qwell has karma of 10 |
21:57.58 | leifmadsen | infobot: Qwell+10 |
21:58.22 | p3nguin | infobot: qwell++ |
21:58.32 | p3nguin | ~karma qwell |
21:58.32 | infobot | qwell has karma of 11 |
21:58.53 | p3nguin | infobot: qwell-- |
21:58.56 | p3nguin | ~karma qwell |
21:58.56 | infobot | qwell has karma of 10 |
21:59.00 | p3nguin | BAM |
21:59.33 | leifmadsen | weirdos |
22:00.08 | p3nguin | smacks leifmadsen around a bit with a rusty iPhone |
22:00.20 | leifmadsen | get a new iPhone? |
22:01.16 | p3nguin | I wish. I'd like to have an iPhone 4 CDMA. |
22:01.44 | p3nguin | I guess I could just buy one, huh? |
22:01.58 | p3nguin | Not sure why I didn't think of that earlier. |
22:04.19 | leifmadsen | I'd prefer to not have an iPhone. |
22:04.22 | leifmadsen | Mission Accomplished! |
22:04.33 | [TK]D-Fender | abhors fruit-based technology |
22:04.52 | carrar | Just get a 1way pager |
22:07.14 | gordonjcp | yup |
22:07.31 | gordonjcp | get an amateur radio licence and homebrew a POCSAG solution |
22:07.34 | gordonjcp | simple |
22:09.15 | gordonjcp | tweak the receiver in your pager to work on 2m or 70cm depending on if it started life on 153 or 466MHz, knock up an encoder, and get it on the air |
22:13.08 | *** join/#asterisk jkroon (~jkroon@dsl-241-236-19.telkomadsl.co.za) |
22:17.24 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
22:17.37 | *** join/#asterisk edge (~edge@97-64-216-2.client.mchsi.com) |
22:17.43 | *** join/#asterisk nmjnb (~nmjnb@213.114.116.245) |
22:17.48 | p3nguin | Why would you need an amateur license for either of those frequencies? Neither is in the ham bands. |
22:18.15 | edge | Is there a way for VoiceMailMain() to grab the extension that is calling it? or somehow get that information to the VoiceMailMain() function? |
22:18.33 | citywok | ${CALLERID(num)} is a good way |
22:18.35 | p3nguin | Extensions don't make calls, phones do. |
22:18.44 | citywok | assuming the callerid of the phone is the same as the mailbox number |
22:18.55 | citywok | otherwise use a database variable to store and look up that info |
22:18.59 | p3nguin | And phone have caller ID numbers, which can coincide with mail box numbers. |
22:19.45 | p3nguin | You can also use accountcode. |
22:21.32 | edge | p3nguin, accountcode? I assume there is a some "prefered" method for doing this. I mean the phone has a button with a mailbox on it, i assume that it can be used for jumping right to the sets mailbox |
22:22.15 | p3nguin | For my Messages key, it is programmed to call *86 (*VM)... |
22:22.36 | p3nguin | And *86 runs VoiceMailMain(${CDR(accountcode)}@default) |
22:22.54 | p3nguin | but it could instead run VoiceMailMain(${CALLERID(num)}@default) |
22:22.58 | edge | do I need to set the accountcode in the sip config? |
22:23.02 | p3nguin | Yes. |
22:23.05 | p3nguin | Or callerid |
22:23.11 | p3nguin | Depending on which method you choose. |
22:23.29 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
22:23.30 | p3nguin | And the accountcode or callerid number will need to be the same as the voice mail box number. |
22:24.31 | edge | p3nguin, I think i will do that, see the 'accountcode' in the sip.conf for the extension, then adjust the dialplan accordingly. |
22:24.44 | p3nguin | extensions are not found in sip.conf |
22:25.26 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
22:26.54 | edge | p3nguin, err i mean peer devices? |
22:27.00 | p3nguin | I'm sure you do. |
22:27.25 | edge | p3nguin, Key systems are the only think i've ever known, it has been so tough to not think of phones as ONLY extensions |
22:27.41 | p3nguin | *shrug* |
22:27.47 | p3nguin | To me, a phone is a phone. |
22:29.02 | p3nguin | I really think it's nap time. |
22:29.38 | edge | p3nguin, Thanks again for the help |
22:31.25 | akrohn | NAPS FOR EVERYONE YAY |
22:32.07 | citywok | yea... i'm on can of coke #2... need energy |
22:33.28 | leifmadsen | I think of phones as devices rather than extensions |
22:33.41 | leifmadsen | then you apply the concept of an extension to the device |
22:34.35 | citywok | leifmadsen: what naming convention do you use for your devices? |
22:34.40 | leifmadsen | you can then abstract a person (or persons) to an extension |
22:34.44 | leifmadsen | citywok: mac address |
22:34.58 | citywok | i was guessing that was the answer heh |
22:35.07 | leifmadsen | but what about the softphones?! |
22:35.17 | leifmadsen | guess what, they utilize a network card that also has a mac address :) |
22:35.35 | citywok | :P |
22:35.46 | citywok | do you have people log in to their phones? |
22:35.50 | *** join/#asterisk PoWeRKiLL (~powerkill@77.125.87.36) |
22:36.17 | citywok | we simply use extensions that follow the people, and i do the no-no thing and think of phones as extensions :P |
22:36.18 | p3nguin | I would only do that for people who do not have their owns desks. |
22:37.01 | citywok | if you move to another PC the softphone knows your extension and that becomes you. not by the book but it works :P |
22:39.30 | p3nguin | The phones do *not* know your extension. |
22:39.40 | p3nguin | They know nothink. nothink. |
22:40.04 | p3nguin | Extensions just dial to devices. The device does not know what extension was used to reach it. |
22:41.33 | p3nguin | If you move to another PC, the soft phone configuration does not change. You just change the extension in asterisk and that's the end of the move. Additionally, most people have some extension information associated with each device in sip.conf settings such as callerid, description, accountcode, mailbox, so you'll want to update those as well. |
22:41.47 | *** join/#asterisk jkroon (~jkroon@dsl-241-250-126.telkomadsl.co.za) |
22:42.08 | p3nguin | But the phones themselves need not be altered in any way. |
22:42.48 | citywok | when an agent launches the softphone it is launched with their personal extension, and that's what the softphone uses to register. it has no clue that is the extension, but it is :) |
22:42.50 | *** part/#asterisk cbwest (~cbwest@nat/cisco/x-pqefvpvtpbjogkla) |
22:43.13 | p3nguin | Again, phones do not know extension information. |
22:43.21 | p3nguin | They just don't. |
22:43.32 | citywok | yes... i'm aware of that. which is why i said "it has no clue that is the extension, but it is " |
22:43.33 | p3nguin | All they know is the user id and the password. |
22:43.56 | citywok | the phone doesn't know that the userid is the extension, nor does it really care. |
22:44.16 | p3nguin | Why would the user ID match the extension number? That's just... silly. |
22:44.52 | citywok | if it works, how is it silly? |
22:45.03 | p3nguin | Extension 762 dials a device by the name of 0123AAAFFFF. The phone only knows that it is a device by the name of 0123AAAAFFFF, not anything to do with the extension. |
22:45.24 | citywok | it works, it's fairly logical, it does the job. |
22:45.36 | p3nguin | It's illogical when you have to made adjustments such as the ones we're discussing. |
22:46.06 | p3nguin | You end up with a phone named 1001 with an extension of 1004, and extension 1004 calls a device named 1007, etc. |
22:46.09 | p3nguin | all wrong. |
22:46.26 | citywok | ah, yea, i don't have that problem at all |
22:46.31 | p3nguin | ~devicenames |
22:46.31 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
22:46.35 | citywok | that would be _craziness_ to manage |
22:47.03 | citywok | in hindsight i would have abstracted it. if i could go back a couple years to when i built this system and teach a couple year younger me what i know now it probably would have been helpful. |
22:47.35 | gordonjcp | so where do the extension numbers and device names match up? |
22:47.45 | p3nguin | This is why we must educate the next generation to not use device names which match extension numbers. |
22:47.46 | citywok | gordonjcp: most people use astdb to look it up |
22:47.53 | p3nguin | gordonjcp: extensions.conf or some db |
22:48.02 | p3nguin | The association is made in the Dial(). |
22:48.40 | citywok | my people are my extensions, and my extensions are applied to my devices directly rather than through a middleman |
22:49.08 | p3nguin | And then additionally, people associate extension information to the device by using settings such as callerid, mailbox, and voicemail. |
22:49.10 | citywok | so i can't hot-desk a physical device with this setup. it requires reassigning the physical devices. |
22:49.16 | p3nguin | s/voicemail/mailbox/ |
22:49.42 | p3nguin | wait... |
22:49.45 | p3nguin | That's not right, either. |
22:49.48 | citywok | fortunately the limitations i accidentally gave myself haven't been a problem |
22:49.51 | p3nguin | callerid, mailbox, and accountcode. |
22:50.16 | citywok | good thing -- it's too late. lol. |
22:50.22 | p3nguin | Good thing you don't care to hot desk. |
22:50.23 | [TK]D-Fender | [17:49]citywokso i can't hot-desk a physical device with this setup. it requires reassigning the physical devices. <- How so? |
22:50.42 | [TK]D-Fender | Device name and "extension" aren't inherently tied to each other... |
22:50.54 | gordonjcp | p3nguin: I am in a worse place for this than being new to it, because I last used asterisk around 1.4 days |
22:50.59 | citywok | [TK]D-Fender: yea, they aren't... unless you do like i did... |
22:51.11 | p3nguin | I used 1.4 until a month ago. |
22:51.19 | [TK]D-Fender | I fail to see why you can't adapt in-place |
22:51.20 | gordonjcp | p3nguin: possibly even earlier than that, I'm just going by some config files kicking around |
22:51.24 | p3nguin | The concepts are still the same. |
22:51.36 | citywok | [TK]D-Fender: i probably could if i really needed to, at the moment it's "if it aint broke don't fix it" |
22:51.39 | gordonjcp | p3nguin: right, but I haven't done it for a couple of years and lots of stuff *has* changed |
22:51.49 | p3nguin | Not too much, really. |
22:51.58 | [TK]D-Fender | citywok: Oh... well "don't really care right now" is another matter. Carry on then... |
22:52.00 | citywok | it only takes a couple seconds to move an extension from one device to another in the GUI we built |
22:52.07 | p3nguin | A few settings here and there, a few apps' syntax. |
22:52.57 | citywok | [TK]D-Fender: That's pretty much what it comes down to. Risking breaking something to solve a problem I don't have isn't something I want to do. I don't want to have to tell my boss i was trying to fix a non-issue when i broke everything and took the call center down. |
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22:53.22 | p3nguin | I'd make the change one device at a time. |
22:53.30 | p3nguin | But that's just how I do things. |
22:54.30 | p3nguin | Today, Jim's phone is getting a new device name. His extension will remain the same, but his phone is going to be reconfigured. |
22:54.32 | citywok | I'm honestly not sure what all I would have to change to unwind this problem. The tools that generate the MAC.cfg files, the scripts that handle agent logins/logouts, the dialers |
22:54.38 | p3nguin | Tomorrow, I'll choose someone else's phone to change. |
22:55.03 | citywok | everything is 100% autoprovisionsd and network driven so I can't really walk around and change one device at a time |
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22:55.21 | gordonjcp | is there a standard set of examples for configuring 1.8? |
22:55.25 | citywok | it's definitely doable, it would just take some serious planning |
22:55.32 | gordonjcp | everything I've seen seems to use numbers as the username |
22:55.33 | citywok | ~thebook |
22:55.33 | 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://www.asteriskdocs.org/ or see ~buybook |
22:55.35 | p3nguin | gordonjcp: There are sample files. |
22:55.45 | p3nguin | ~device names |
22:55.46 | gordonjcp | p3nguin: yeah, but I find them incomprehensible |
22:55.47 | citywok | gordonjcp: see the book |
22:55.47 | p3nguin | ~devicenames |
22:55.47 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
22:55.56 | gordonjcp | citywok: I've no way of getting that |
22:56.03 | p3nguin | You have no internets? |
22:56.04 | citywok | it's in pdf format. |
22:56.07 | gordonjcp | citywok: I'm a very long way from a bookshop |
22:56.09 | p3nguin | It's also in html. |
22:56.23 | citywok | or that :) |
22:56.32 | gordonjcp | oh, okay |
22:57.10 | citywok | or ask russel or leif if they have any copies of it left :p |
22:57.57 | p3nguin | The key to this is to comprehend that there is no relationship between phone and extension until you (the admin) creates one through the use of dial plan apps. |
22:58.11 | p3nguin | And don't tie a phone to a person. |
22:58.22 | p3nguin | Give a person an extension. It will be his extension for life. |
22:58.27 | citywok | gordonjcp: listen to p3nguin he speaks the truf. and as he says, don't make the mistake i did and tie them together. it's a bitch to unwind later on. |
22:58.45 | p3nguin | Then associate any random device of your choosing with the extension. |
22:59.56 | p3nguin | My extension is 762. It will be mine no matter which office I sit in, which phone I use in said office, and no matter how many phones I break and/or dispose of. |
23:00.47 | p3nguin | If I move around often, I'll hot-desk, and I'll have to login when I arrive at any random phone. |
23:01.11 | p3nguin | If I move around only sometimes, I'll change the device that my extension Dial()s. |
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23:07.47 | gordonjcp | hm, I only have one adaptor for my ata |
23:09.47 | gordonjcp | invents |
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