00:00.23 | p3nguin | http://pastebin.com/tER2jGnY |
00:00.33 | SeRi | awesome. Thanks |
00:04.19 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
00:08.01 | SeRi | for the iax.conf acl I use my brothers ip under allowed correct? |
00:08.29 | SupYoshi | F2Knight :D Working on the readme now, I just want to say im a complete Idiot and you were absolutly right on every single line :P Thank you very much just needed a littl push here |
00:09.13 | p3nguin | deny=0.0.0.0/0.0.0.0 |
00:09.14 | p3nguin | permit=209.16.236.73/255.255.255.0 |
00:09.30 | p3nguin | where permit contains his address and netmask. |
00:09.40 | SeRi | awesome thanks! |
00:09.51 | p3nguin | If his address never changes, a mask of 255.255.255.255 would be best. |
00:10.02 | *** join/#asterisk wdoekes2 (~walter@wjd.osso.nl) |
00:10.20 | SeRi | can hosts be used? |
00:10.23 | p3nguin | no |
00:10.27 | SeRi | ok. |
00:10.57 | p3nguin | But since his address is static, you don't need to worry with a name anyway. |
00:11.17 | SeRi | got it. I was wondering for something like callcentric.com |
00:12.37 | p3nguin | They suck. |
00:13.47 | SeRi | indeed |
00:13.52 | SeRi | :( |
00:14.24 | SeRi | I cant seem to let them go for some bull shit reason... :( |
00:16.02 | p3nguin | You've got a dirt cheap DID with them, so I understand. |
00:16.24 | p3nguin | But to keep them means that you will put up with their crap. |
00:17.30 | SeRi | tell me about it. you are been nice by saying *crap*.... |
00:18.49 | *** join/#asterisk vinhdizzo (~vinh@pool-74-100-182-10.lsanca.fios.verizon.net) |
00:19.13 | *** join/#asterisk wdoekes2 (~walter@wjd.osso.nl) |
01:10.37 | p3nguin | We finally got 100 Mbit cable service available to the consumer. It has been testing in the lab for many months. |
01:13.41 | *** join/#asterisk coppice (~coppice@m121-202-3-210.smartone.com) |
01:16.17 | SeRi | nice |
01:16.24 | SeRi | 3.? |
01:16.29 | SeRi | 3.0* |
01:29.25 | *** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:5:0:558:12) |
01:33.02 | *** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:5:0:558:12) |
01:40.08 | *** join/#asterisk TheBigS (~TheBigS@c-69-255-105-123.hsd1.va.comcast.net) |
01:40.43 | TheBigS | Anyone have any experience setting up Polycom 331 <--- complete asterisk noob |
01:46.32 | *** join/#asterisk sixohquad (~Cody@184.65.142.249) |
01:51.11 | sixohquad | hey guys, is anyone able to help me out by telling me more about the errors im receiving here? im trying to dial out when this happens, was working fine, stopped working now and delivers that series of errors |
01:51.14 | sixohquad | http://pastebin.com/kNaxgquw |
01:53.10 | WIMPy | Before you broke it? The sip peer seems to be missing. |
01:54.18 | sixohquad | ok |
01:55.38 | *** join/#asterisk SupYoshi (SupaYoshiL@ip51cc8577.speed.planet.nl) |
01:56.18 | SupYoshi | Hi :) Does anyone know good dutch voices for asterisk? |
01:57.01 | sixohquad | mmm, 0 sip registrations |
01:57.05 | sixohquad | wtf |
01:58.30 | *** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:5:0:558:12) |
02:10.47 | *** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:5:0:558:12) |
02:20.04 | sixohquad | im receiving other errors now. http://pastebin.com/E1Qbrgr3 |
02:20.17 | sixohquad | that is my sip.conf, extensions.conf, and the errors are at the bottom |
02:21.39 | [TK]D-Fender | sixohquad: Your sip.conf has no [general] section and is completely broken because of it |
02:21.48 | sixohquad | thank you :) |
02:22.22 | [TK]D-Fender | sixohquad: I also se no register statements at all... so I don't see why you would expect even a hope of seeing an attempt let alone a success in "sip show registry" |
02:22.32 | [TK]D-Fender | sixohquad: that is for* registering outbound like to an ITSP, etc |
02:22.40 | [TK]D-Fender | sixohquad: not to check phones registering to * |
02:22.43 | sixohquad | i had one in there |
02:22.48 | sixohquad | i had a sip register line |
02:22.48 | [TK]D-Fender | "had" |
02:22.51 | [TK]D-Fender | maybe |
02:22.54 | sixohquad | lol |
02:22.57 | sixohquad | kthx |
02:23.12 | [TK]D-Fender | first [general] then registrations, THEN your peers |
02:23.15 | [TK]D-Fender | order matters |
02:23.20 | [TK]D-Fender | missing bits = fail |
02:26.09 | carrar | hey |
02:26.14 | carrar | isn't there a book that covers this? |
02:26.16 | [TK]D-Fender | I'm off for a bit... |
02:26.26 | [TK]D-Fender | ~osmosis |
02:26.26 | infobot | [~osmosis] Osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
02:26.28 | [TK]D-Fender | ;) |
02:26.34 | carrar | heh |
02:26.45 | carrar | ~book |
02:26.45 | 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 |
02:33.04 | sixohquad | i have the book |
02:33.06 | sixohquad | im just a n00b |
02:33.18 | sixohquad | i have it open beside me |
02:33.38 | sixohquad | but when your a n00b l0zer sometimes you dont know what to look for to solve your problem :) |
02:35.19 | *** join/#asterisk Yourname` (~Yourname@unaffiliated/yourname/x-837320) |
02:38.29 | sixohquad | should the register line go in the general context? |
02:40.31 | *** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:5:0:558:12) |
02:41.07 | p3nguin | It is required to be in the general settings section. |
02:42.28 | F2Knight | SupYoshi, np, glad you got it. |
02:47.14 | sixohquad | ok, now im running into this: http://pastebin.com/431zKZ9U |
02:47.17 | sixohquad | thanks btw p3nguin |
02:50.08 | *** join/#asterisk luckman212 (~do-not-re@pool-108-41-8-176.nycmny.fios.verizon.net) |
02:55.35 | p3nguin | Your ITSP probably require you to be registered before you can make calls. |
02:58.21 | sixohquad | i am registered now |
02:58.46 | sixohquad | state: registered |
02:59.12 | p3nguin | Without adequate SIP debug, there's not much else I can guess about. |
03:00.29 | *** join/#asterisk LostyJai (~blah@202.171.190.130) |
03:02.02 | sixohquad | what would you like me to do that will help you more? |
03:02.38 | p3nguin | core set verbose 3 |
03:02.39 | p3nguin | sip set debug on |
03:02.40 | p3nguin | Make a call, pastebin the entire output. |
03:03.25 | p3nguin | To turn sip debug back off, sip set debug off |
03:06.40 | sixohquad | http://pastebin.com/5ArEJqg1 |
03:07.15 | LostyJai | hey all |
03:07.29 | LostyJai | got a weird problem, when i dial 990 to log an agent on or off |
03:07.54 | *** join/#asterisk roxdragon (~gianni@unaffiliated/roxdragon) |
03:07.56 | roxdragon | hi all... |
03:08.05 | roxdragon | how to load ALL module on asterisk? |
03:08.07 | LostyJai | when it says enter a new extension, i type in the agent's extension... it'll say agent is logged in, then "I'm sorry, that's not a valid extension" |
03:08.21 | roxdragon | No such command 'sip show peers' (type 'core show help sip show' for other possible commands) |
03:12.23 | p3nguin | roxdragon: What does your modules.conf look like? |
03:12.52 | p3nguin | sixohquad: I don't see any problem indicated in your pastebin. It looks like a successful call to me. |
03:15.36 | *** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:5:0:558:12) |
03:16.04 | roxdragon | p3nguin, solved... but |
03:16.05 | roxdragon | ERROR[2560]: rtp_engine.c:253 ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? |
03:16.18 | p3nguin | roxdragon: What does your modules.conf look like? |
03:16.59 | roxdragon | p3nguin, http://paste.ubuntu.com/797842/ |
03:17.26 | roxdragon | how to load all modules? |
03:18.01 | p3nguin | Line 8 of that file takes care of doing that. All you have to do is provide a conf for each module. |
03:18.44 | roxdragon | yes but don't load |
03:19.04 | p3nguin | What does your rtp.conf look like? |
03:19.19 | roxdragon | 2 modules loaded |
03:19.21 | roxdragon | one moment |
03:19.48 | roxdragon | http://paste.ubuntu.com/797845/ |
03:20.24 | p3nguin | That one also looks okay. Have you tried restarting asterisk recently? core restart now |
03:20.59 | roxdragon | that's ok |
03:20.59 | roxdragon | :D |
03:21.02 | roxdragon | thank u |
03:21.34 | roxdragon | it's work |
03:21.41 | roxdragon | :D thank youuuuuu goodnight |
03:24.02 | LostyJai | http://pastebin.com/7zGup75U |
03:24.06 | LostyJai | this is the log of my error |
03:25.38 | p3nguin | There is no extension by the name of 'return' in that context. |
03:25.49 | p3nguin | line 8 |
03:26.21 | LostyJai | how do i fix that? |
03:26.43 | p3nguin | Create the extension. |
03:26.54 | LostyJai | called return? |
03:26.55 | LostyJai | ok |
03:26.57 | LostyJai | thanks |
03:27.26 | p3nguin | I'm more concerned with the reason the call is being sent to that extension in the first place. |
03:27.39 | LostyJai | no idea, just did a restore |
03:27.48 | LostyJai | which config file would that be set in |
03:27.50 | LostyJai | and what should ti be |
03:28.09 | p3nguin | Never heard of "a restore." Sounds strangely like a Windows term. |
03:28.12 | *** join/#asterisk sixohquad (~Cody@184.65.142.249) |
03:28.29 | LostyJai | trixbox |
03:28.31 | sixohquad | p3nguin, it seems to connect the call properly now, but hangs up immediately after connecting. |
03:28.42 | p3nguin | Yeah, we don't support trixbox here. |
03:28.44 | p3nguin | ~trixbox |
03:28.44 | infobot | Trixbox is unable to be supported here. It is a closed-source distribtuion of Asterisk and FreePBX containing proprietary code that its users do not have access to, making it difficult to support. Try joining #trixbox and asking your questions there. |
03:30.14 | LostyJai | thought agent login/logoff was an asterisk thing |
03:30.17 | LostyJai | but okay, thanks |
03:30.44 | *** join/#asterisk brah (c82b2429@gateway/web/freenode/ip.200.43.36.41) |
03:30.45 | p3nguin | It is, but it's too bad that YOU are not in control of asterisk. |
03:31.06 | LostyJai | yeah i don't know my asterisk |
03:31.23 | LostyJai | system was build before my time, was just given to me |
03:31.28 | p3nguin | There's a book for that. |
03:31.30 | p3nguin | ~book |
03:31.31 | 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 |
03:33.05 | sixohquad | p3nguin, do you see any indication in the debug as to why it would hang up the call on me right away/ |
03:33.34 | p3nguin | I didn't notice anything like that. What I saw was a successful call. |
03:34.18 | sixohquad | hm, strange |
03:34.22 | p3nguin | Have you made calls to asterisk which stop at asterisk? Have you made calls through your ITSP which originate at asterisk and not from a phone? |
03:35.25 | sixohquad | i can call other extensions internally just fine |
03:35.32 | sixohquad | how do i originate a call from asterisk? |
03:36.00 | sixohquad | make an extension that dials that number? |
03:36.01 | p3nguin | On the CLI, use the "channel originate" command. |
03:36.09 | sixohquad | ok will give it a shot |
03:36.11 | p3nguin | You obviously already have the extension. |
03:36.20 | p3nguin | Otherwise it would not have made the call that you showed me. |
03:36.23 | sixohquad | yeah |
03:36.24 | sixohquad | true |
03:36.43 | sixohquad | dont quite have the full concept of dialplans yet, but i do see what you're saying |
03:37.49 | p3nguin | channel originate SIP/call-labs/1604-your-number application Playback tt-weasels |
03:43.30 | sixohquad | ughhh, putty keeps jumping my scrolling and im having trouble copy/pasting it because of it lol |
03:44.21 | p3nguin | I'm not sure what you mean. If you can better describe the issue with PuTTY, maybe I can help you solve that so you can continue troubleshooting. |
03:44.50 | sixohquad | when i try to scroll back, something else hits the CLI and makes me jump back to the bottom of the log |
03:45.17 | p3nguin | sip set debug off, core set verbose off |
03:45.21 | sixohquad | i cant nice |
03:46.04 | p3nguin | *shrug* |
03:46.30 | sixohquad | oops, i only meant to type nice\ |
03:46.31 | sixohquad | sorry |
03:48.30 | sixohquad | ugh, i damn near pasted it all in here by accident |
03:48.32 | sixohquad | http://pastebin.com/nVPa6RkJ |
03:50.01 | p3nguin | Looks successful. |
03:50.22 | sixohquad | it didn't call my phone at all though? |
03:50.40 | p3nguin | It had to have done something. |
03:50.45 | sixohquad | lol |
03:50.59 | sixohquad | everything except call me :) |
03:51.26 | p3nguin | I'm not seeing the verbose messages in with the sip debug. |
03:52.09 | p3nguin | If it's there, you'll have to point it out. If it isn't there, you'll have to fix that. |
03:52.24 | sixohquad | it should have played back the weasels thing shouldn't it? |
03:52.35 | p3nguin | It played it. |
03:52.50 | p3nguin | Line 184 says so. |
03:53.15 | p3nguin | But if I had the verbose messages in here like I originally asked for, I could see the dialplan progression. |
03:53.44 | *** join/#asterisk Zert (~zert@l49-17-157.cn.ru) |
03:53.46 | Zert | hellp |
03:53.49 | Zert | hello |
03:53.50 | sixohquad | im sorry i must have missed you asking for a verbose messaGE |
03:54.07 | Zert | is there any method to read DTMF in AGI during STREAM FILE execution? |
03:54.13 | p3nguin | (2102.37) <p3nguin> core set verbose 3 |
03:54.13 | p3nguin | (2102.39) <p3nguin> sip set debug on |
03:54.13 | p3nguin | (2102.40) <p3nguin> Make a call, pastebin the entire output. |
03:54.23 | sixohquad | oh, its on verbosity 9 |
03:54.25 | sixohquad | you want me to lower it? |
03:54.35 | p3nguin | 9 is just silly. |
03:54.46 | Zert | I don't want to wait until playback completed |
03:54.51 | sixohquad | oh ok lol |
03:55.00 | p3nguin | But 9 would still show me the messages... which I do not see. |
03:55.11 | p3nguin | If they are there, I need you to point them out. |
03:55.59 | p3nguin | Wait... |
03:56.05 | p3nguin | Disregard. |
03:56.16 | p3nguin | You aren't using any dialplan in this test. |
03:56.23 | p3nguin | That's why it doesn't appear. |
03:56.29 | sixohquad | http://pastebin.com/SWMw2GG3 is the new pastebin |
03:56.52 | p3nguin | You're connecting directly to SIP/your-itsp/your-number without dial plan. |
03:56.53 | resist0r | haha 9 is just silly |
03:57.27 | sixohquad | p3nguin, im not sure i understand, how should i do it differently? |
03:58.04 | p3nguin | The test does not get much more basic than this. |
03:58.33 | p3nguin | Asterisk originates a call to the destination specified. When there is an answer, it runs the application specified. |
04:00.06 | sixohquad | yeah, i understand that, i thought thats what you were doing with the originate command? |
04:00.19 | p3nguin | That's what I'm talking about. |
04:00.27 | sixohquad | whenever i make a call from the softphone, it says connection established, and then says call ended |
04:00.33 | p3nguin | Note the part about the answer. |
04:01.00 | p3nguin | If the channel does not go into state "Up" (answered), the application should never execute. Yours executed. |
04:02.08 | sixohquad | http://pastebin.com/ZfKANcsx |
04:02.15 | sixohquad | there's my extensions.conf |
04:02.36 | p3nguin | This originate does not utilize extensions. |
04:03.25 | sixohquad | so what should i do? |
04:03.34 | p3nguin | Your default context is pretty useless there. |
04:03.35 | sixohquad | how do you suggest i work through this? |
04:03.47 | sixohquad | i copied that context out of the book ;) |
04:03.49 | p3nguin | Find out how your phone is answering the call. |
04:03.54 | sixohquad | its not |
04:03.56 | sixohquad | guaranteed |
04:03.59 | sixohquad | its calling my cellphone |
04:04.04 | p3nguin | It is. Guaranteed. |
04:04.04 | sixohquad | and it doesn't even ring |
04:04.11 | sixohquad | no, there is something else going on |
04:04.13 | p3nguin | Because: (2201.00) <p3nguin> If the channel does not go into state "Up" (answered), the application should never execute. Yours executed. |
04:04.15 | sixohquad | my phone is not answering the call |
04:04.26 | p3nguin | There's an answer. |
04:04.40 | p3nguin | Otherwise the app would not execute. |
04:04.52 | sixohquad | fantastic |
04:04.58 | sixohquad | so there's something answering the call, and its not me |
04:05.07 | p3nguin | Call some other number. |
04:07.35 | sixohquad | phone never rang |
04:07.37 | sixohquad | http://pastebin.com/bPD5hPV2 |
04:07.42 | sixohquad | called a different number |
04:10.41 | p3nguin | I've had another idea. |
04:10.52 | p3nguin | What is this call-labs thing? |
04:12.13 | sixohquad | just a context |
04:12.16 | sixohquad | sip trunk |
04:12.18 | p3nguin | uh no |
04:12.23 | p3nguin | ~trunk |
04:12.23 | infobot | somebody said trunk was a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
04:12.54 | sixohquad | ok, my sip provider? |
04:12.58 | Yourname` | If a SIP trunk is actually a GSM gateway - how does the DTMFing situation work if the GSM gw isn't sending proper DTMF tones? |
04:12.59 | p3nguin | Is it an ITSP? Is it something of yours? A company? Need Details! |
04:13.11 | sixohquad | ITSP :) |
04:13.15 | sixohquad | sorry i dont know what details to give yet |
04:13.17 | sixohquad | i'll get there |
04:13.18 | sixohquad | lol |
04:13.43 | p3nguin | My thought is that they are poorly configured and they are answering calls when they should not be. |
04:13.50 | sixohquad | its vdi |
04:14.04 | p3nguin | I don't know what that is. |
04:14.11 | sixohquad | oh ok |
04:14.30 | sixohquad | its a company |
04:14.35 | *** part/#asterisk Zert (~zert@l49-17-157.cn.ru) |
04:14.47 | p3nguin | Can you use another provider for a test? |
04:14.56 | sixohquad | sure |
04:17.57 | sixohquad | omg |
04:17.59 | sixohquad | it works fine |
04:18.05 | sixohquad | shakes his head |
04:18.18 | sixohquad | i'll find out whats going on with that provider tomorrow |
04:18.26 | p3nguin | Okay, so your provider is teh suck. |
04:18.36 | sixohquad | yeah, something's configured wrong |
04:18.46 | p3nguin | They are not supposed to be answering your calls that you are sending out. |
04:19.08 | sixohquad | yeah thats no good |
04:19.46 | sixohquad | im sure its just a configuration error |
04:19.54 | p3nguin | I'm wondering if it would ever make it out ever. |
04:19.59 | sixohquad | yeah |
04:20.07 | sixohquad | i think the billing panel has just given me some wrong information |
04:21.23 | sixohquad | anyways, thanks for your help, sorry for running you in circles needlessly, you can imagine how bad it was confusing a nub :) |
04:21.34 | p3nguin | Try another test through call-labs. Replace "Playback tt-weasels" with "Dial SIP/1001" |
04:21.58 | sixohquad | through the provider that didn't work?\ |
04:22.07 | p3nguin | You tried this already? |
04:22.14 | sixohquad | no |
04:22.22 | p3nguin | I'll wait. |
04:23.41 | p3nguin | As soon as asterisk gets the answer indicator, your SIP phone should ring. Go ahead and pick it up and wait a few seconds to see what happens next. I'm curious what they are doing with your calls after they incorrectly answer. |
04:24.44 | sixohquad | ok, this provider hasn't given me a proper register line i dont think, if i exclude it, will that ruin everything? |
04:24.48 | sixohquad | i have a context for it |
04:25.07 | p3nguin | You can't have a "context" for it. Contexts are something else. |
04:25.45 | p3nguin | Registration is for telling the other side how to reach you. Additionally, many providers will require you to register first before you are allowed to send calls inbound to them. |
04:29.33 | sixohquad | hrm |
04:29.35 | sixohquad | you know what |
04:29.41 | sixohquad | i think i need to talk to this provider tomorrow |
04:29.44 | sixohquad | and see what is happening |
04:30.13 | sixohquad | after we sort out how its supposed to be properly configured, i'll debug like this again and see if my calls get through |
04:30.21 | sixohquad | i think its likely a config problem, im probably missing something |
04:30.27 | p3nguin | When you tried that last test, did you end up talking to yourself? |
04:30.34 | sixohquad | the fact that they haven't provided a register line could be the key to all of this |
04:30.39 | sixohquad | no |
04:31.09 | p3nguin | Remember, registration is to tell them how to reach you so they can send calls to you. |
04:31.38 | sixohquad | ok |
04:31.47 | sixohquad | so outgoing should work fine without it |
04:32.10 | p3nguin | Unless they require registration first, outgoing has nothing to do with registration. |
04:32.23 | p3nguin | But you said earlier that it was registered successfully, anyway. |
04:34.21 | sixohquad | http://pastebin.com/SDt0ejC1 |
04:34.48 | sixohquad | yeah, its complicated and we're not too worried about incoming calls at this point so i'll confirm that part with my ITSP tomorrow |
04:35.52 | sixohquad | there are multiple servers, and i dont think i was given the right information |
04:36.03 | p3nguin | That gave a forbidden response. THAT is what I would expect to see from a provider who requires registration before you can send calls in. |
04:36.10 | sixohquad | ok |
04:36.13 | sixohquad | so i need that register line |
04:36.19 | *** join/#asterisk bandroidx (~bandroidx@2607:f358:1:fed5:5:0:558:12) |
04:36.42 | p3nguin | In addition to that, your application probably would have failed to run if the call would have been accepted. |
04:38.27 | sixohquad | ok, i'll check with them tomorrow |
04:39.39 | sixohquad | annoying to find that this was caused by improper information, but im glad to know my asterisk works |
04:39.43 | sixohquad | with the other provider |
04:39.59 | sixohquad | i'll go back to the book now :) |
04:40.09 | sixohquad | thanks again for your help p3nguin, i really appreciate your time |
04:41.07 | p3nguin | I hope it works out for you. Let me know what they have to say about answering your outgoing calls. |
04:41.53 | sixohquad | yeah i'll let you know |
04:41.58 | sixohquad | do you know chigambamukoko |
04:42.00 | sixohquad | ? |
04:42.08 | sixohquad | irc nick |
04:43.11 | sixohquad | i gotta go get some grocery shopping done. ttyl, thanks again |
04:43.36 | luckman212 | i'm curious about Asterisk's TRANSFER_CONTEXT |
04:46.10 | luckman212 | there seems to be no mention of it in "the" ~book |
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05:45.50 | [TK]D-Fender | luckman212: And "The Book" can't include absolutely everything without requiring a forklift |
05:47.26 | Maliuta | [TK]D-Fender: if it came with a forklift that'd be cool! ;) |
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07:41.25 | F2Knight | Q: Does anyone know a way to tell what functions/applications/commands are loaded by what modules? Some are easy but other modules load several additional tools. |
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07:43.49 | schmidts | good morning |
07:44.01 | F2Knight | morning schmidts |
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07:47.22 | kaldemar | F2Knight: egrep -r "function name=|application name=" * |
07:47.42 | IsUp | hello |
07:47.55 | kaldemar | F2Knight: the source files have XML documentation for the functions and applications they provide. |
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07:51.14 | F2Knight | kaldemar, should I run that in the source dir? |
07:52.25 | F2Knight | kaldemar, nvm running it in source gave what I was looking for .. Thanks! |
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07:56.54 | mirelab | Q: Does anyone know if queue member status shouldn't change if member of a queue is for example SIP/333@<ip> or SIP/333@<peer defined in sip.conf> |
07:57.36 | MariusAgon | Morning, guys. I have to send an e-mail from my dialplan, doing it with System() app. Maybe it's more linux CLI based question, but how can I break line in my e-mail body? Thank you in advice. |
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07:58.48 | F2Knight | mirelab, sorry but can you explain your question better? |
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07:59.41 | F2Knight | MariusAgon, System() will execute your shell statement.. (I like mutt as an email client) \n is the standard new line char for linux systems |
07:59.50 | mirelab | F2Knight: I've posted it on Asterisk::forum => http://forums.digium.com/viewtopic.php?f=1&t=81197&sid=97269b2de52a547237f4203bfb7dbd36 |
08:00.29 | MariusAgon | F2Knight thank you. |
08:00.44 | F2Knight | MariusAgon, you may want to consider some more thought in the logic of that call though.. You may find creating a simple shell or python script that you can pass variables to will work better to populate a templated email |
08:02.17 | mirelab | F2Knight: can you take a look on forum> i can explain here too if needed :) |
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08:04.15 | F2Knight | Your member line looks wrong... |
08:04.32 | F2Knight | you have SIP/4003@4003 |
08:04.51 | F2Knight | That would dial extension 4003 using the sip account 4003 |
08:05.15 | F2Knight | your member should just be SIP/4003 |
08:06.35 | F2Knight | or you may want to have the extensions in a context ... say [myqueued-agents] |
08:06.55 | F2Knight | and then do a Local/4003@myqueued-agents/n |
08:07.39 | F2Knight | the /n is magic on a LOCAL channel it maked the calls paths flow like other channels,.. normially local does not stay in the call paths. |
08:07.58 | mirelab | F2Knight: i tried with SIP/4003 what was logic to me too, but than I always get "busy noone has answerd..." |
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08:08.33 | F2Knight | are you trying to dial from the same phone? |
08:09.38 | mirelab | F2Knight: no, just want that [4003] to be dialed agent that will ring when member is in queue |
08:09.58 | mirelab | F2Knight: but status get stuck on Ringing |
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10:07.25 | dj_hamsta | is there something similar to Digium Academy - Asterisk Essentials Online Video Training Course 6 month but free? |
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10:12.06 | MariusAgon | Hello, guys. Trying to execute an php script from my dialplan, but seems i can't pass or catch variables(I'm poor scripter). From my dialplan i'm passing them like this : System(php /usr/queue.php ${number} ${time} ${queue}) (thoese set before in dialplan). Is this correct way? Thank you in advice. |
10:13.28 | jacc0 | @MariusAgon: why are you using System() and not AGI? |
10:13.43 | MariusAgon | I'm not very familliar with AGI :/ |
10:13.54 | jacc0 | just rename your php to agi |
10:13.56 | jacc0 | :) |
10:14.50 | MariusAgon | Hmz, going to try that, thank you jacc0! :) |
10:14.53 | ollii | you dont need to rename it |
10:14.58 | ollii | Agi(foo.php,123) |
10:15.07 | ollii | location should be var/lib/asterisk/agi-bin/ |
10:15.12 | jacc0 | put this line in top of your php: |
10:15.17 | jacc0 | #!/usr/bin/php |
10:15.18 | ollii | #!/usr/bin/php |
10:15.20 | ollii | yeah |
10:15.31 | ollii | if this is your php binary path |
10:15.32 | jacc0 | ;) |
10:15.45 | MariusAgon | Ok,thank you, going to try that. |
10:16.04 | jacc0 | I need to re-wire my pc; bye, ttyal |
10:16.09 | MariusAgon | bye |
10:16.20 | ollii | MariusAgon: http://www.voip-info.org/wiki/view/Asterisk+AGI |
10:16.27 | ollii | that might help you |
10:17.24 | MariusAgon | OK, thanks |
10:22.33 | mirelab | Can anyone help me with queue member status, => http://forums.digium.com/viewtopic.php?f=1&t=81197&sid=dd4647da750735dceeb67dca9eb16578 |
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10:26.56 | MariusAgon | ollii: still can't catch sent arguments, trying to do that with $argv[1] etc. And getting error "ast_carefulwrite: write() returned error: Broken pipe". Maybe i still miss something.. |
10:31.23 | mirelab | MariusAgon: try to catch response to a new variable which you don't need to use after that, that can remove Broken pipe error sometimes |
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10:33.25 | ollii | MariusAgon: http://pastebin.com/ePvvFgnz |
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10:34.32 | ollii | this is simple |
10:34.46 | ollii | there already exists a phpagi class: http://phpagi.sourceforge.net/ |
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10:46.40 | DanFromUK | hi, is there a quick way to end a sip channel from the cli? |
10:47.12 | DanFromUK | i can see the channel using core show channels, its an old channel that didnt end for some reason, but its blocking calls because of call-limit |
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10:49.33 | ollii | depends on your version...soft hangup [TAB] |
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10:52.15 | DanFromUK | thanks. that did the job |
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11:31.27 | gego | Hi. Is there a way to get a QuadBRI PCI card to work with dahdi? I have one runnning with misdn_v1 on an antique kernel but have to update this now. |
11:32.13 | gego | I've tried with misdn_v2 which works but only in combination witch lcr - which I want to avoid |
11:32.23 | WIMPy | Depends on what you do with it. You might need a patch or you can use misdn v2. |
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11:33.25 | WIMPy | didn't like the ide of an extra application, but it has been the most stable solution so far. |
11:34.03 | WIMPy | Although I just discovered a bug in the transmission of dialling information between lcr and Asterisk. |
11:34.05 | gego | WIMPy: I use it as an interface to euro-isdn - and to an analog fax - so both TE and NT mode are needed |
11:34.39 | WIMPy | Does your card have 4 or 5 jumpers per interface for TE/NT selection? |
11:35.20 | gego | let me have a look. Oh no wait: let me have a cigarette instead ... |
11:37.03 | tzafrir_laptop | gego, which card? |
11:37.27 | tzafrir_laptop | what does dahdi_hardware (from dahdi-tools) have to say about that card? |
11:37.56 | tzafrir_laptop | If you don't have dahdi-tools installed and don't want to fully install it, |
11:38.13 | tzafrir_laptop | download the latest version and run from the source tree: ./xpp/dahdi_hardware |
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11:46.14 | gego | tzafrir_laptop: it's a BeroNet BN8S0 |
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11:47.09 | tzafrir_laptop | gego, so I suppose dahdi / chan_dahdi should support it |
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11:47.57 | itsurkg | hello all, is there a repository for asterisk 1.8. I need to install asterisk1.8 with gui |
11:48.03 | gego | I installed 2.6.0-rc1 - is that the latest? |
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11:49.14 | joobie | hey guys.. i usually use polycom cos the audio quality is great.. but i need a phone that has about 8 extensions (without adding an additional module to the polycom phones) |
11:49.40 | joobie | what would be the next best phone in terms of audio ? |
11:49.51 | gego | tzafrir_laptop, dahdi_hardware tells me: driver should be 'qozap' but is actually 'wcb4xxp' \n pci:0000:03:06.0 qozap+ 1397:16b8 Generic OctoBRI ISDN card |
11:49.54 | joobie | need to look at other brands cos polycom dont do one ofthose phones unless you buy their LED ones |
11:49.56 | joobie | which are pricey |
11:50.12 | tzafrir_laptop | hmm... "generic" means it is probably not detected |
11:50.20 | WIMPy | joobie: Snom 3x0 allow 12 accounts and have G.722 support. |
11:50.25 | tzafrir_laptop | Most likely will require a trivial patch |
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11:51.05 | joobie | good audio WIMPy ? |
11:51.30 | WIMPy | joobie: G.722 is wideband. |
11:51.50 | joobie | I dont use that |
11:51.54 | joobie | I use g729 |
11:51.57 | joobie | low compression |
11:52.06 | joobie | how does g722 compare in terms of b/w |
11:52.20 | gego | WIMPy, it's got 4 jumpers each. |
11:52.27 | WIMPy | 64kbps just like G.711 |
11:52.52 | joobie | ahh |
11:52.59 | WIMPy | gego: In that case you will need the te_nt_override patch for wcb4xxp to get NT mode. |
11:53.10 | joobie | ergh |
11:53.11 | joobie | gtg |
11:53.14 | joobie | brb later |
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11:53.36 | tzafrir_laptop | WIMPy, are you sure? The card has no jumpers? |
11:54.07 | tzafrir_laptop | (feel free to comment on https://issues.asterisk.org/jira/browse/DAHLIN-203 ) |
11:54.11 | WIMPy | tzafrir_laptop: He said only 4. wcb4xxp needs the famous fifth jumper. |
11:54.55 | tzafrir_laptop | uses here a Junghanns card with just 4 |
11:55.10 | tzafrir_laptop | (for testing, of course) |
11:55.27 | WIMPy | tzafrir_laptop: he, I uploaded the updated patches there :-) |
11:56.24 | tzafrir_laptop | IIRC the problem with the patch there is that it breaks configuration with jumpers |
11:56.30 | WIMPy | Without the patch you cannot set NT mode in software. It will only try to read the state of that fifth jumper. |
11:56.42 | tzafrir_laptop | This is why that patch is on hold |
11:56.56 | gego | WIMPy, where do I find these patches and to what do I have to apply em? |
11:57.18 | WIMPy | From what I read the configuration by jumper should still work. But I can't test that. My cards only have 4. |
11:58.30 | WIMPy | The configuration ist very ugly anyway. You have to configure the same thing in three different places. |
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11:58.59 | WIMPy | Either (Jumper or module parameter), dahdi/system.conf and asterisk/chan_dahdi.conf. |
11:59.30 | WIMPy | So getting rid of at least one of the three places would be nice. |
12:00.01 | WIMPy | gego: Follow the link tzafrir_laptop posted. The latest patch still works, even with some offset. |
12:00.03 | gego | maybe I've just blacklisted/inserted the wrong modules (or maybe just in the wrong order) -> why does dahdi_hardware say: driver should be qozap ? |
12:01.32 | WIMPy | qozap is an older addon. Not sure how current that is. |
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12:11.59 | WIMPy | has a feeling that WaitExten is eating the digits. |
12:13.14 | schmidts | WIMPY WaitExten is allways hungry ;) |
12:13.42 | WIMPy | yes, but I can see the digits arrive, but the exten does not include all of them. |
12:15.52 | schmidts | WIMPy how can you see it? dtmf debug? |
12:16.13 | WIMPy | "dtmf", yes. |
12:16.47 | WIMPy | tries to add additional debug output. |
12:23.30 | WIMPy | Ugh. Does someone know how WaitExten works? It seems to exig after the first digit is received. Where is the rest received? |
12:29.33 | schmidts | WIMPy imho you can give a parameter how many extension you want to get |
12:30.25 | WIMPy | The work seems to be done somewhere else. |
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12:36.33 | dj_hamsta | i need a quick guide on how to add users and setting up sip phones within asterisk now 1.7 |
12:36.44 | dj_hamsta | i found an old one but the config is different |
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12:38.49 | fobus912 | Hi All |
12:39.03 | mirelab | Deas anyone know why queue member status is not changing from (Ringing) if member is SIP/<ext>@<sip_peer> or (Unknown) if member is SIP/<ext>@<IP adress> and could it be changed? |
12:39.06 | fobus912 | Does anybody know how to configure the dial plan to send DTMF after the call is bridged |
12:39.31 | jacc0 | @dj_hamsta: edit sip.conf and ad a phone. edit extensions.conf and add a dialplan for that extension |
12:39.31 | fobus912 | I have configured the dialplan as the following |
12:39.32 | fobus912 | exten => _1XXX,1,Dial(SIP/${EXTEN:1}@FXO,,D(wAwwwwBwCwDwAwBwCwDwAwB)) |
12:39.42 | fobus912 | but the DTMF are sent before the call is bridged |
12:39.52 | dj_hamsta | jacc0, i am using the gui portion freepbx |
12:40.05 | jacc0 | then you are in the wrong channel |
12:40.39 | jacc0 | join #freepbx |
12:40.41 | fobus912 | can please someone advise the correct configuration |
12:43.02 | jacc0 | @fobus912: create a contex in extensions,conf like: extens = _X.,1,sendDTMF(abcdabcdabc) |
12:43.21 | jacc0 | and then use originate to connect a phone to this dialplan when it picks up |
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12:45.18 | jacc0 | originate(sip/${EXTEN},exten,yoursenddtmfcontext,${EXTEN},1) |
12:46.58 | fobus912 | Let me try this jacc0 Thanks for the suggestion |
12:47.26 | kaldemar | fobus912: maybe with U() or M() that uses app SendDTMF. D() will send the digits between answer and bridge, so that won't work for you. |
12:49.28 | mirelab | Deas anyone know why queue member status is not changing from (Ringing) if member is SIP/<ext>@<sip_peer> or (Unknown) if member is SIP/<ext>@<IP adress> |
12:50.24 | fobus912 | kaldemar i just tried with Macro and usign M is not working |
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12:56.43 | fobus912 | jacc0 can't get it working |
12:56.46 | fobus912 | as you suggested |
12:58.52 | fobus912 | is there any other way i can achieve it please |
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13:02.47 | jacc0 | it works for sure; please show me your extensions.conf |
13:03.12 | jacc0 | and your cli output |
13:03.13 | jacc0 | ~pb |
13:03.13 | 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. |
13:04.59 | fobus912 | jacc0 here is my extension conf |
13:05.00 | fobus912 | [test] |
13:05.00 | fobus912 | extn=_1XX,1,SendDTMF(ABCD) |
13:05.15 | fobus912 | originate(sip/${EXTEN:1}@FXO,exten,test,${EXTEN:1}@FXO,1) |
13:05.42 | fobus912 | *originate(sip/${EXTEN:1}@FXO,extn,test,${EXTEN:1}@FXO,1)* |
13:06.07 | fobus912 | I'm really not sure how i should configure it |
13:07.29 | jacc0 | originate(sip/${EXTEN:1}@FXO,extn,test,${EXTEN:1},1) |
13:07.33 | jacc0 | next time |
13:07.35 | jacc0 | ~pb |
13:07.35 | 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. |
13:07.55 | jacc0 | so i can edit it :) |
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13:10.56 | fobus912 | jacc0 is it the correct way to do it ? |
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13:22.48 | jacc0 | originate(sip/${EXTEN:1}@FXO,extn,test,${EXTEN:1},1) |
13:23.12 | mirelab | Deas anyone know why queue member status is not changing from (Ringing) if member is SIP/<ext>@<sip_peer> or (Unknown) if member is SIP/<ext>@<IP adress> |
13:23.14 | mirelab | <PROTECTED> |
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13:27.21 | jacc0 | you should remove the second @FXO |
13:30.08 | BenC[UK] | Can anyone give me names of any tier1 uk voip carriers? |
13:30.12 | [TK]D-Fender | mirelab, Do not expect member status that way. use a Local channel to do the dial instead |
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13:31.13 | hrolf | How do I get the wait time for a call abandoned in queue? |
13:31.40 | [TK]D-Fender | hrolf, Its in the queue log |
13:32.08 | mirelab | [TK]D-Fender: Local/.../n with n to be normal channel, so i was told, but still too bad that status is not changing :( |
13:32.27 | hrolf | [TK]D-Fender: Sorry I failed to mention that I need it through the Manager API. |
13:32.33 | [TK]D-Fender | mirelab, It probabkly will once you change |
13:33.07 | hrolf | Like the Leave event (when a call leaves a queue) only reports the call ID, channel etc. But not the wait time. |
13:33.21 | [TK]D-Fender | hrolf, Then you alredy knew your answer |
13:33.45 | hrolf | [TK]D-Fender: Does that mean I can get it? |
13:33.50 | hrolf | s/can/can't |
13:34.06 | [TK]D-Fender | hrolf, No, it means you can already see that it doesn't give it to you the way you want. |
13:34.08 | mirelab | [TK]D-Fender: i think i worked with Local member and as i can remeber it's working fine, but i hate adding contexts with 1 line for a feature like that lol |
13:34.41 | hrolf | [TK]D-Fender: Fine. So is there something that I can do about it? |
13:36.19 | BenC[UK] | hrolf, can't you monitor events and keep track of time that way? |
13:36.25 | [TK]D-Fender | hrolf, You've got the source.... |
13:38.31 | hrolf | BenC[UK]: Yep, I was just asking if there is any already built solution for this thing. I think I'll have to do it my self by keeping track of the events. |
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14:00.12 | qakhan | hi all |
14:00.58 | qakhan | i have 4 agents in a queue, when call comes in and no agent take the call then all agents logoff automatically |
14:01.05 | qakhan | plz help |
14:01.54 | [TK]D-Fender | qakhan, read the sample queues.conf . It lists the features that will kick agents who don't answer |
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14:15.44 | qakhan | [TK]D-Fender can u tell me? |
14:21.40 | *** join/#asterisk jacco_7564 (~Jacco@195-241-134-176.ip.telfort.nl) |
14:21.44 | jacco_7564 | Hello |
14:22.36 | jacco_7564 | the database asterisk is using, is that mysql? |
14:23.40 | Gugge | if you set it up to use mysql yes |
14:23.45 | [TK]D-Fender | jacco_7564, what "database"? |
14:23.59 | schmidts | jacco_7564 what asterisk version? |
14:24.10 | [TK]D-Fender | jacco_7564, Only DB * inherently uses is AstDB and that is either BDB pre *10 or SQLite |
14:24.10 | leifmadsen | jacco_7564: the built in AstDB is a key/value pair DB based on Berkeley DB |
14:24.19 | leifmadsen | in Asterisk 10 and later, the backend is SQLite |
14:24.24 | leifmadsen | but it operates the same right now |
14:24.32 | jacco_7564 | Using asterisk 1.8 |
14:24.38 | leifmadsen | then it is berkeley DB |
14:24.43 | [TK]D-Fender | jacco_7564, Then BDB |
14:25.03 | jacco_7564 | Ok, thanks! |
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14:48.55 | *** join/#asterisk Assalino (53f49842@gateway/web/freenode/ip.83.244.152.66) |
14:49.07 | Assalino | Hey hey |
14:49.19 | Assalino | total Asterisk noob here |
14:49.35 | Assalino | I'm following a tutorial so I can setup an Asterisk server with FreePBX |
14:49.46 | WIMPy | ~freepbx |
14:49.47 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
14:49.47 | Assalino | which seems to be popular amongst the ASterisk devs |
14:50.08 | Assalino | thanks WIMPy. My question isn't related to FreePBX, I think |
14:50.12 | Assalino | I just wondered |
14:50.16 | Assalino | once i set it up |
14:50.20 | WIMPy | GUIs certainly aren't popular in here. |
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14:50.26 | Assalino | can I make a call from the service? |
14:50.41 | Assalino | I'm not sure if ASterisk supports call making out-of-the-box |
14:50.42 | WIMPy | What service? |
14:50.49 | Assalino | from the server, sorry |
14:51.07 | [TK]D-Fender | From the server to where? |
14:51.13 | [TK]D-Fender | Doing what? |
14:51.15 | WIMPy | You have to configure at least one peer and at least one extension to make a call. |
14:51.17 | [TK]D-Fender | When? |
14:51.33 | Assalino | well, ideally what I want to do is create a service where users can input their phone numbers and get a call from ASterisk |
14:51.39 | Assalino | that plays a recorded message |
14:51.41 | [TK]D-Fender | Assalino, Yes you can do this |
14:51.46 | WIMPy | Hmm. No only an extension if you accept guests. |
14:51.59 | Assalino | right now I'm just playing with it to see if I can build a prototype |
14:52.10 | [TK]D-Fender | Assalino, And so far FreePBX does virtually nothing to assist this process |
14:52.26 | Assalino | but my guess is that I'd need to subscribe to some form of service to be able to make calls, no? |
14:52.36 | [TK]D-Fender | Assalino, naturally |
14:52.52 | Assalino | is this something Asterisk provide, or do you use a 3rd party? |
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14:53.06 | [TK]D-Fender | Assalino, we call them "The Telephone Company" |
14:53.07 | Assalino | I read somewhere you could do it via Skype and Google Voice, but I'm not sure if it's the same |
14:53.10 | WIMPy | You need some sort of connection to the PSTN. Either on some sort of line or via an ITSP. |
14:53.15 | [TK]D-Fender | Assalino, And no, Asterisk is not the telco |
14:53.32 | Assalino | so you can't do it purely via VoIP? |
14:53.38 | [TK]D-Fender | ~itsp |
14:53.39 | 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. |
14:53.40 | [TK]D-Fender | ^^^ |
14:53.59 | [TK]D-Fender | Yes, and "Asterisk" is not a service provider. It is software that can talk to one however |
14:54.06 | Assalino | I see |
14:54.20 | Assalino | I knew that you could get hardware to connect an asterisk server to phone lines |
14:54.27 | Assalino | but I thought we could use VoIP otherwise |
14:54.33 | Assalino | which seems ideal to me |
14:54.34 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
14:54.43 | sekil | digium b410p is fully functional today? no need for misdn? |
14:55.29 | [TK]D-Fender | sekil, Been DAHDI compaible for years AFAIK |
14:55.34 | WIMPy | sekil: Sort of. Depending on your card and needs, you might need a patch. |
14:56.03 | WIMPy | Or is it specifically about the b410p? |
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14:56.20 | [TK]D-Fender | WIMPy, Were there multiple hardware releases of that card for which a patch is required? |
14:56.27 | *** join/#asterisk fobus912 (~fobus912@41.143.31.230) |
14:56.31 | [TK]D-Fender | WIMPy, Otherwise he did just tell you the model.. |
14:57.01 | Assalino | @[TK]D-Fender, so if I set up a server, without having the hardware to connect to the ITSP, I won't be able to make calls? |
14:57.20 | WIMPy | I wan't sure if the question was about the card or the driver. As the obvious choice for that card would be dahdi. |
14:57.30 | [TK]D-Fender | Assalino, If your server has internet connetivity then you have the "hardware" |
14:57.50 | [TK]D-Fender | Assalino, And you need something to make calls. What that is depends on what you want to use |
14:57.50 | Assalino | yup, that's what I thought |
14:58.05 | Assalino | right, so now I need to find a VoIP provider in the UK then |
14:58.08 | WIMPy | I've been told the b410p does not work reliably with misdn2. |
14:58.09 | Assalino | that works with ASterisk |
14:58.15 | [TK]D-Fender | ~itsplist-uk |
14:58.15 | infobot | extra, extra, read all about it, itsplist-uk is UK based ITSps include http://www.voiptalk.org/ http://www.voipon.co.uk/ http://www.gradwell.com/ and a few other tinpot companies you can dig up with google. |
14:58.58 | Assalino | awesome |
14:59.00 | Assalino | you guys rule :) |
14:59.54 | Assalino | thank you |
15:00.34 | sekil | I'm still on * 1.2 and misdn |
15:00.56 | WIMPy | Ouch |
15:01.53 | sekil | hm...looks like libpri is all I need nowadays.. |
15:02.34 | WIMPy | misdn1 is also still maintained. |
15:02.39 | WIMPy | for some reason. |
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15:03.20 | sekil | I'd rather go for libpri and chan_zap/dahdi if possible |
15:03.35 | sekil | misdn needed a lot of kernel patching etc. |
15:04.13 | WIMPy | You don't need to patch, but the old version is no longer included, either. |
15:04.16 | *** part/#asterisk irroot (~gregory@197.170.58.6) |
15:04.26 | sekil | old version? |
15:04.34 | sekil | misdn1 you mean |
15:04.36 | sekil | ? |
15:04.42 | WIMPy | yes |
15:04.51 | sekil | you mean misdn is in the kernel nowadays |
15:05.05 | sekil | ? |
15:05.15 | WIMPy | It has been for may years. |
15:05.31 | sekil | brushes the spider web |
15:05.34 | WIMPy | misdn1 up to 2.6.25.4 and misdn2 since 2.6.26. |
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15:06.11 | sekil | hm... |
15:07.04 | sekil | on 2.6.21 I had to patch |
15:07.08 | sekil | good to know |
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15:07.44 | WIMPy | Unless you needed special features or some hardware ID update, you shouldn't have needed to patch. |
15:08.15 | voipeng | hello, we turned up an additional server and it is replying to register requests on port 3:3 .... any idea where there is a static assignment for all 5060 traffic in asterisk? |
15:08.32 | WIMPy | Or actually, you never need to patch, as you can build the modules standalone, just like you would do with dahdi. |
15:08.43 | voipeng | tcpdumps show it try to connect but the port connection is refused as its not a real registration port |
15:10.03 | voipeng | http://pastebin.com/0pXd4H2d |
15:10.19 | sekil | yeah true...but I'm pretty sure misdn wasn't in 2.6.21 so I had to do something.. |
15:11.27 | sekil | dahdi doesn't support hfc-s single port cards also as I recall.. |
15:11.46 | WIMPy | Not without 3rd party addons, no. |
15:12.13 | WIMPy | And no USB support at all. |
15:12.15 | sekil | yah |
15:12.22 | sekil | pci ones.. |
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15:19.32 | fobus912 | Hi all |
15:20.03 | fobus912 | Is there anyone who can tell me if it's possible to send DTMF with SendDTMF application after the call is bridged |
15:20.20 | fobus912 | using the D() option does send the DTMF but before the call is bridged |
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15:23.51 | jacco_7564 | On outbound routes, is the prefix or prepend used to let asterisk know that it is dialing outside? for example i want to have to dial 00, and then a normal telephone number |
15:27.41 | kaldemar | jacco_7564: prefixes are one choice. read up on dialplan patterns and dialplan structure in general to get an idea on how decisions can be made in dialplan. |
15:28.23 | ollii | gego: ask beronet ... they've a live support on beronet.com ... very friendly... |
15:30.23 | [TK]D-Fender | jacco_7564, #freepbx <- |
15:30.24 | gego | ollii: beronet have a very friendly support for their berofix cards/boxes, but not for the pci-cards which have reached EOL ... for them ... |
15:30.36 | ollii | oh okay :( |
15:31.20 | WIMPy | Luckily most of them are the same. |
15:31.48 | ollii | by now they're using hfc chips on their cards...maybe this old card is also hfc based |
15:36.13 | voipeng | any suggestions why our registration server would reply like this? http://pastebin.com/0pXd4H2d |
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15:43.43 | r0m|u | is dijib in jail? |
15:44.32 | Qwell | who? probably. |
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15:45.11 | r0m|u | lol I havent seen him arround for a while.... all ways talking about getting high and what not.... |
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16:32.43 | p3nguin | r0m|u: Maybe he's in jail for recording all his phone calls without consent. |
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16:41.10 | r0m|u | p3nguin: lol |
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17:01.26 | roxdragon | hi all.. i recive this message when arrive incoming call |
17:01.26 | roxdragon | dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
17:01.31 | roxdragon | what's this? |
17:01.43 | Qwell | roxdragon: more context needed |
17:02.23 | roxdragon | can you help me? |
17:02.34 | Qwell | not without more context, no |
17:03.10 | p3nguin | I would say that the phone you are trying to Dial() is not available. Show us the sip debug. |
17:03.32 | roxdragon | Qwell, extensions.conf? |
17:04.07 | fobus912 | hi all |
17:04.12 | roxdragon | the phone is registred |
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17:04.22 | p3nguin | Show us the sip debug. |
17:04.29 | p3nguin | core set verbose 3 |
17:04.33 | p3nguin | sip set debug on |
17:04.34 | fobus912 | can please someone advise how i can cheive the following, Dial DTMF from the dial plan after the channel is bridged |
17:04.38 | p3nguin | Make a call, pastebin the entire output. |
17:05.08 | fobus912 | i was able to do it with D() but it dial the DTMF before the channel is bridged |
17:05.29 | r0m|u | man we have a nasty storm down here |
17:05.58 | fobus912 | i'm fighting all day long to make it work |
17:06.07 | fobus912 | and i can't find the solution for this |
17:07.40 | roxdragon | ok |
17:08.32 | fobus912 | anyone please ? |
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17:22.45 | *** join/#asterisk odin917 (~Gavin_Sil@66.246.94.130) |
17:23.06 | odin917 | Hi All! I am having an issue with bandwidth.com SIP trunks for incoming calls (outgoing works fine). when i call into my DID for my pbx i can see the traffic hitting my server (via tcpdump -s0 -n -v udp port 5060 and host 216.82.224.202) but i get no output in asterisk CLI. |
17:23.07 | odin917 | here is the tcpdump wher ei can see the traffic http://pastebin.com/TFgHbA5c |
17:23.48 | [TK]D-Fender | <fobus912> i was able to do it with D() but it dial the DTMF before the channel is bridged <- I seriously doubt that. Show us. |
17:23.57 | [TK]D-Fender | ~pb |
17:23.57 | 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. |
17:26.48 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
17:27.01 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
17:27.10 | SuperNull | Hey guys, anyone know of a commercial grade CDR billing software? |
17:38.35 | Qwell | Does "commercial grade" mean it has to be Java, and crash a lot? |
17:39.04 | carrar | I means it only runs in a DOS window |
17:39.25 | [TK]D-Fender | SuperNull, MS Excel |
17:39.27 | WIMPy | A Macro for Excel would be fine, I guess. |
17:39.31 | Qwell | ha |
17:47.29 | *** join/#asterisk hdiogenes (~humberto@189.124.171.215) |
17:48.56 | *** part/#asterisk hdiogenes (~humberto@189.124.171.215) |
17:49.17 | *** join/#asterisk sekil (~sekil@78.24.104.82) |
17:52.48 | paulc | SuperNull: How about A2Billing - have you looked at that? |
17:53.44 | *** join/#asterisk Troyblackman_ (~samuelmat@65-36-42-216.static.grandenetworks.net) |
17:54.15 | Troyblackman_ | hello |
17:55.19 | Troyblackman_ | anyone in here? |
17:56.00 | [TK]D-Fender | Nope. |
17:56.09 | carrar | shhh |
17:57.05 | p3nguin | Now look what you've done. |
17:57.11 | p3nguin | Now he's going to ask things. |
17:57.11 | *** join/#asterisk sekil (~sekil@78.24.104.82) |
17:58.16 | Troyblackman_ | Ill just use forums |
17:58.49 | [TK]D-Fender | lol |
17:59.12 | [TK]D-Fender | Or if you have a question you could just ask.... |
17:59.57 | Troyblackman_ | im just stuck on getting my did to accept incoming calls |
18:00.13 | Troyblackman_ | ill research more on my own and see what I can figure out |
18:00.20 | [TK]D-Fender | Troyblackman_, PASTEBIN the failed attempt so we can see what's wrong |
18:00.22 | [TK]D-Fender | ~pb |
18:00.22 | 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. |
18:00.24 | [TK]D-Fender | ^^^ |
18:00.54 | Troyblackman_ | ok |
18:01.58 | *** join/#asterisk sekil (~sekil@78.24.104.82) |
18:01.58 | Troyblackman_ | Ill be back later on after I do a bit more research and if I cant figure out I will pastebin it |
18:02.04 | Troyblackman_ | Thanks again |
18:03.07 | *** part/#asterisk Troyblackman_ (~samuelmat@65-36-42-216.static.grandenetworks.net) |
18:04.51 | drmessano | lol |
18:15.11 | jaytee | I keep getting this message, "Windows has encountered an error and needs to close". I can't tell if it's an error or a feature though. |
18:16.33 | [TK]D-Fender | jaytee, Ones that turn your screen blue are SPECIAL FEATURES |
18:19.58 | p3nguin | Oh, I see the problem. |
18:19.59 | *** join/#asterisk slidesinger-lt (~jtatum@c-174-57-5-70.hsd1.nj.comcast.net) |
18:20.21 | p3nguin | *Windows* has encountered ... |
18:28.11 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
18:28.11 | *** mode/#asterisk [+o file] by ChanServ |
18:29.55 | *** join/#asterisk akrohn (~akrohn@38.101.60.42) |
18:30.57 | r0m|u | lol |
18:31.26 | WIMPy | Windows _was_ encountered. |
18:36.12 | *** join/#asterisk longhorn (~ady@netaware.ro) |
18:37.05 | longhorn | hi. I have a asterisk sistem going for 2 years now. I just changed the voice provider and now I get the following notice in logs: [Jan 9 20:36:47] NOTICE[2894] chan_sip.c: Registration from '<sip:1002@192.168.111.1>' failed for '192.168.111.3' - No matching peer found |
18:37.24 | longhorn | and I can;t receive calls only place calls outside |
18:37.42 | longhorn | can someone help me with some advices |
18:37.45 | longhorn | thanks |
18:37.56 | beek | longhorn: Did you change sip.conf? |
18:38.11 | longhorn | beek no i change nothing |
18:38.19 | beek | Well, you're going to need to do so. |
18:38.23 | longhorn | just the cables from one device to another |
18:38.31 | longhorn | please guide me |
18:38.47 | longhorn | a friend did the configs |
18:38.48 | WIMPy | What exactely did you change? |
18:38.50 | beek | What do you mean by "voice provider"? |
18:38.51 | longhorn | and i don;t know much |
18:39.05 | longhorn | from one company to another |
18:39.26 | WIMPy | Companies of/for what? |
18:39.29 | beek | Well, you'll need credentials for the new company and make the appropriate changes in sip.conf to register with them. |
18:39.32 | longhorn | cable from one provider device to another nothing to do with the astersk |
18:39.53 | longhorn | i have access to all system |
18:40.00 | longhorn | tell me what I have to do ? |
18:40.02 | WIMPy | Ar you talking about phone lines? Or internet connection? Or what? |
18:40.03 | beek | longhorn: look in sip.conf for the old registration information. You'll need to replace it with the new. |
18:40.17 | longhorn | WIMPy phone lines |
18:41.00 | WIMPy | Ok, that shouldn't matter. Any other changes? |
18:41.12 | longhorn | WIMPy none |
18:41.17 | WIMPy | Like e.g. new router or something? |
18:41.25 | longhorn | beek what in /etc/asterisk/sip.conf ? |
18:41.29 | longhorn | no |
18:41.45 | beek | longhorn: What exactly did you change? Your telco provider? |
18:41.48 | p3nguin | sip.conf isn't going to be involved in the phone line configuration... |
18:41.53 | WIMPy | Changing phone lines will not affect your SIP clients. There must have been another change. |
18:42.01 | longhorn | beek yes |
18:42.02 | p3nguin | However, the NOTICE you received is regarding your SIP phone. |
18:42.12 | *** join/#asterisk jkroon (~jkroon@dsl-241-253-235.telkomadsl.co.za) |
18:42.16 | longhorn | let me tell you mi configuration |
18:42.18 | beek | longhorn: And these are real analog POTS lines and not a VoIP provider? |
18:42.32 | longhorn | it is a VOIP provider |
18:42.36 | longhorn | i'm from Romania |
18:42.39 | longhorn | it is called UPC |
18:42.55 | WIMPy | So NOT phone lines? |
18:43.27 | longhorn | the cable that gets in MP-114 FXO it is normal phone cable |
18:43.40 | beek | WTF? |
18:43.50 | longhorn | 192.168.111.3 device is a MP-114 FXO |
18:43.56 | *** join/#asterisk oej (~olle@87.96.134.129) |
18:44.08 | longhorn | beek maibe i can't explain this very right ... |
18:44.46 | WIMPy | Ah, so somehting on that device has changed, apart from the other cable connected to it, it seems. |
18:44.46 | p3nguin | You don't need to have your gateway device registering to asterisk. Configure it for static. |
18:45.25 | beek | longhorn: Let's start again. Disregard the company for a moment... are you connecting to the PSTN via POTS (analog line) or via sip? |
18:45.26 | longhorn | p3nguin and how i do that ? |
18:45.54 | p3nguin | Do you have access to the device? |
18:46.34 | longhorn | p3nguin yes i have |
18:46.37 | p3nguin | If you do not have access, asterisk may have to be changed instead. I would prefer to configure the device, though. |
18:46.56 | longhorn | beek sip |
18:47.45 | beek | longhorn: Pastebin your sip.conf file (***** out any password information) |
18:48.54 | longhorn | http://pastebin.com/kZmGAkj6 |
18:49.42 | beek | Crap.... FreePBX. You'll need to pastebin all of those included files as well longhorn. |
18:50.05 | WIMPy | Or go to #freepbx |
18:50.06 | p3nguin | ~freepbx |
18:50.06 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
18:50.12 | p3nguin | We don't do FreePBX. |
18:50.15 | p3nguin | And we don't do Windows. |
18:50.31 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
18:50.42 | longhorn | p3nguin i have linux |
18:50.51 | r0m|u | pukes |
18:51.22 | beek | longhorn: pastebin those include files so I can see how you're actually connecting. |
18:51.39 | p3nguin | I suppose you aren't familiar with the phrase. "I don't do windows" means that I do not wash window glass. I was merely playing with the phrase. |
18:51.42 | p3nguin | Carry on. |
18:52.42 | p3nguin | needs to call Deutschland. |
18:52.43 | longhorn | http://pastebin.com/1DBq6Fpi |
18:53.14 | p3nguin | I suppose it is too late to be in the office, though. |
18:53.30 | p3nguin | I should have called them at 6 am here. |
18:53.31 | WIMPy | 19:53 |
18:53.36 | WIMPy | Probably |
18:53.43 | p3nguin | 12:53 here |
18:54.25 | beek | longhorn: Pastebin sip_registrations_custom.conf and sip_registrations.conf |
18:54.48 | longhorn | beek they are empty |
18:55.27 | beek | How about sip_custom.conf and sip_additional.conf? |
18:55.30 | beek | hates FreePBX |
18:57.48 | longhorn | http://pastebin.com/veAebDJX |
18:58.13 | longhorn | beek only sip_additional.conf has content |
18:59.18 | beek | brb |
19:00.26 | *** join/#asterisk netman (netman@214.162.76.188.dynamic.jazztel.es) |
19:03.05 | adeel|work | i'm experiencing call quality issues terminating calls to a few NPA's and was looking for an alternative SIP provider, than my current one (voip.ms)....any suggestions? |
19:03.26 | adeel|work | preferably US based |
19:03.28 | p3nguin | flowroute |
19:10.37 | *** join/#asterisk bandroidx (~bandroidx@205.185.117.159) |
19:16.56 | *** join/#asterisk troyt (~troyt@c-24-10-222-127.hsd1.ut.comcast.net) |
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19:22.13 | *** join/#asterisk gurra (~gurra__@unaffiliated/gurra) |
19:22.19 | beek | longhorn: Just to confirm... you moved analog phone cables from one place to another and made no other changes to anything? |
19:23.07 | longhorn | beek right |
19:23.34 | longhorn | that is why I can;t explain ... and I don;t know much about this sistem |
19:24.12 | beek | And to confirm... it's <asterisk> --- <MP-114> --- PSTN ? |
19:25.08 | longhorn | yes |
19:25.12 | *** join/#asterisk troyt (~troyt@c-24-10-222-127.hsd1.ut.comcast.net) |
19:26.23 | beek | And what part did you physically change? |
19:26.57 | longhorn | beek just the wire |
19:27.20 | beek | From the <MP-114> to the PSTN... |
19:27.22 | beek | ? |
19:27.34 | longhorn | no |
19:27.42 | longhorn | sorry |
19:27.43 | longhorn | tes |
19:27.44 | longhorn | yes |
19:27.56 | WIMPy | maybe, finally... |
19:27.57 | WIMPy | d'oh |
19:28.26 | WIMPy | That alone wouldn't change anythin on the Asterisk side. |
19:28.41 | WIMPy | Somethign else must have happend. |
19:28.59 | beek | Definitely. Something else has changed. |
19:29.15 | [TK]D-Fender | longhorn, pastebin "sip show peers" |
19:30.30 | longhorn | http://pastebin.com/2JzESYi6 |
19:32.15 | [TK]D-Fender | <longhorn> hi. I have a asterisk sistem going for 2 years now. I just changed the voice provider and now I get the following notice in logs: [Jan 9 20:36:47] NOTICE[2894] chan_sip.c: Registration from '<sip:1002@192.168.111.1>' failed for '192.168.111.3' - No matching peer |
19:32.33 | [TK]D-Fender | I know *I* don't see a "1002" in that list anywhere... |
19:33.24 | WIMPy | No, it's 800. |
19:33.27 | [TK]D-Fender | Also odd that so many other's are pointing to the same IP. |
19:33.44 | [TK]D-Fender | that reg attempt doesn't seem to think "800" |
19:35.04 | [TK]D-Fender | Oh.. actually... there is the key bit.. |
19:35.09 | [TK]D-Fender | it CAN'T register. |
19:35.12 | [TK]D-Fender | the host is fixed |
19:35.17 | [TK]D-Fender | and shouldn't be |
19:35.48 | [TK]D-Fender | And fromuser=800 is the wrong parm to make the names match. That is for * calling out on it, |
19:35.52 | WIMPy | Well, maybe that message has always been there. |
19:35.55 | [TK]D-Fender | it should have been "username=800" |
19:36.10 | [TK]D-Fender | Several minor silly breakages |
19:36.23 | [TK]D-Fender | Fix the host, restrict the IP's fix the username |
19:40.45 | *** join/#asterisk pdtpatrick_ (~pdtpatric@12.249.4.226) |
19:41.29 | pdtpatrick_ | Question - is there anyway to see channel count with sip? sip show channels ..something? |
19:42.14 | akrohn | 'core show calls' could be close to what you're looking for |
19:45.00 | pdtpatrick_ | that does it - thanks :) |
19:49.32 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
19:52.00 | *** join/#asterisk netman (netman@214.162.76.188.dynamic.jazztel.es) |
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20:06.21 | p3nguin | I would prefer to configure MY on-premise devices to use static entries in asterisk instead of registrations. Registrations are for things that move around and/or change. |
20:17.13 | *** join/#asterisk nny (~Scott@174.107.223.14) |
20:17.54 | nny | I feel stupid for asking this, but does asterisk have a voicemail feature to allow access when the voicemail message is played as the user (example press * and login to hear messages). I can't recall and the voip-info doc is lacking |
20:18.09 | *** join/#asterisk malcolmd_ (~malcolmd@pdpc/sponsor/digium/malcolmd) |
20:18.09 | *** mode/#asterisk [+o malcolmd_] by ChanServ |
20:18.47 | WIMPy | Yes, look at the a and o extensions. |
20:19.28 | nny | WIMPy: ahh thanks, got it |
20:20.02 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
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20:27.32 | *** join/#asterisk Yourname` (~Yourname@unaffiliated/yourname/x-837320) |
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20:32.12 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
20:37.11 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
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20:40.17 | p3nguin | Is there a sound file that says "You do not need to dial a 1 to call this number"? |
20:53.16 | malcolmd | not that i remember, no |
20:53.36 | malcolmd | each sounds package includes a txt file that includes written descriptions of each prompt |
20:54.20 | *** join/#asterisk resist0r (~resist0r@69.31.131.51) |
21:01.04 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
21:03.09 | *** join/#asterisk jks (~jks@193.189.93.254) |
21:11.02 | p3nguin | I looked there first, and I didn't see it. Then I asked, hoping someone knew something about it that I didn't. |
21:11.38 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
21:12.37 | [TK]D-Fender | p3nguin, A very worthy commision for Allison |
21:13.33 | p3nguin | Also one that says, "There is no need to dial a 9 before the phone number. Please stop doing it." |
21:14.03 | [TK]D-Fender | Keep it up.. you'll hit a volume rebate level ;) |
21:14.14 | p3nguin | I know her hourly rate, but it would only take a few minutes to record a few sentences... does she have a minimum? |
21:15.13 | _Corey_ | Digium used to have a one-off type of product on their website for these kinds of prompts... not sure if it's still around |
21:15.20 | [TK]D-Fender | Dunno... pretty sure her rates are posted by hr site, otherwise fire off an e-mail... |
21:15.30 | p3nguin | Maybe we could get her to do it for the asterisk project and not for me. |
21:15.38 | p3nguin | I'll email her. |
21:16.14 | [TK]D-Fender | p3nguin, If you're worried about usage rights You could hand them over.. but asking her to do it "for the poject" ... this is her living. What she did alerady was billed |
21:28.15 | *** join/#asterisk gurra (~gurra__@unaffiliated/gurra) |
21:31.48 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
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21:41.40 | *** join/#asterisk sixohquad (~sixoh@74.198.150.96) |
21:43.26 | sixohquad | If you guys were doing offsite asterisk pbx's for companies, would you run multiple companies in one asterisk install, or would you virtualize your servers and run one company per virtualization? |
21:44.02 | fenrus | i'd give each company a vm somewhere |
21:44.17 | fenrus | amazon or some other cloudprovider |
21:46.30 | *** join/#asterisk joesuffceren (~chatzilla@ip68-104-167-226.ph.ph.cox.net) |
21:47.54 | joesuffceren | is anyone using a Dell R210ii server with digium hardware (TE121 specifically)? I am having trouble with it (two separate R210ii servers and two separate TE121 cards and no joy) that digium support (thus far) cannot sort out, and Dell support has told me "it's not a supported card" |
21:48.52 | WIMPy | It might help if you tell us what the issue is. |
21:49.10 | joesuffceren | I'm just wondering if it might be software related and/or if anyone else has done it before. I'm running the FreePBX distro on both boxes. I'm thinking of installing asterisk/dahdi/libpri from source as a test. |
21:49.23 | joesuffceren | WIMPy: getting there. :-) |
21:49.48 | *** join/#asterisk sixohquad (~sixoh@74.198.150.96) |
21:50.08 | joesuffceren | the card throws PCI read/write errors in dmesg, and dahdi won't restart (gives an error from dahdi_cfg about span not existing). |
21:50.18 | sixohquad | Sorry got disco. Did anybody answer my previous question? |
21:50.24 | joesuffceren | digium tech SSH'd into the system and said the config is good |
21:50.37 | WIMPy | That sounds evil. |
21:51.46 | fenrus | i have no really clue about theese matters |
21:52.08 | fenrus | but i'd go to my server bios and try disabling stuff i dont need |
21:52.15 | fenrus | and enableing legacy mode |
21:52.16 | fenrus | etc |
21:52.26 | fenrus | could be some "feature" that fucks stuff up |
21:52.29 | p3nguin | It seems much easier to keep things separated if each company has its own asterisk system. |
21:52.37 | WIMPy | Maybe latencies can be configured. |
21:52.54 | WIMPy | Or disable spread spectrum. |
21:53.25 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
21:54.44 | sixohquad | P3Nguin, you're probably right, but the financial side is much more lucrative if you can fill a server with more clients :) |
21:55.21 | sixohquad | Can you include different files from inside sip.conf and extensions.conf? |
21:55.23 | WIMPy | And much easier to offer free calls between them :-) |
21:55.45 | WIMPy | Sure |
21:56.26 | sixohquad | So keeping it clean would be easy. Keep each company in separate files. |
21:56.47 | fenrus | mysql backend perhaps? |
21:56.54 | joesuffceren | WIMPy/fenrus: I'll look into those. thanks. I was just hoping for the silver bullet of someone saying "we had that same problem and resolved it by ___________ |
21:56.57 | joesuffceren | :-) |
21:57.25 | WIMPy | ____________ = Get a cheaper server. |
21:57.28 | p3nguin | You could certainly use one system for multiple companies. I didn't think that was the question. |
21:57.51 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
21:58.13 | sixohquad | Yeah, I was mostly fishing for pro's and cons of doing either way |
21:58.26 | joesuffceren | yeah, corporate policy is to use dell for everything and use their financing, etc., so my hands are tied on that one |
21:58.45 | fenrus | often reffered to as 'dellhell' |
21:58.48 | sixohquad | That makes one hard drive failure epic though |
21:58.54 | WIMPy | So what cards do Dell support? |
21:59.17 | joesuffceren | nothing from digium. They do sell Cisco T1 cards, but I wasn't interested in going there |
21:59.54 | WIMPy | They certainly won;t offer drivers for Asterisk. |
22:02.33 | *** join/#asterisk macroevolve (~macroevol@c-98-234-125-202.hsd1.ca.comcast.net) |
22:02.35 | p3nguin | sixohquad: Sure, but you'll have a nice backup plan and disaster recovery plan to minimize downtime in the event of failure, so that would almost be a non-issue. |
22:04.47 | macroevolve | hey guys - i was curious for anyone that has started a outbound calling company - if one of your telemarketers is calling on behalf of Company A (Customer A) and the prospect want to be put on the DNC list, does this prevent you from calling that same prospect even on behalf of other companies? |
22:07.05 | [TK]D-Fender | macroevolve: I cannot image it would |
22:08.02 | macroevolve | TKD: thx! |
22:08.33 | p3nguin | Maybe if there is a certain significant common factor of the companies... and I don't mean that they use the same phone company. |
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22:12.12 | *** join/#asterisk bbourdage (~bbourdage@h243.138.141.67.dynamic.ip.windstream.net) |
22:12.38 | amaninacan | Hey guys, I've got a dumb question about VoIP in general(let me preface this with the fact that I'm trying to win an arugment) |
22:13.01 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
22:13.01 | *** mode/#asterisk [+o malcolmd] by ChanServ |
22:13.15 | [TK]D-Fender | So what you're saying is you want us to tell you what you want to hear. Gotcha. Go on... |
22:13.37 | sixohquad | P3nguin, thx for the input. Ill keep it in mind as I learn more here |
22:13.37 | sixohquad | Haha |
22:13.41 | amaninacan | Basically, we've got a hosted VoIP provider that says in order to get their shit to work, we have to use port triggering. However, in every experince I've had with VoIP (I've managed setups from 3-300 phones) we've used something like STUN/ICE. |
22:13.48 | amaninacan | Which is preferred? |
22:14.09 | p3nguin | You don't NEED either of those options. |
22:14.25 | p3nguin | I don't use port triggering and I don't use STUN. |
22:15.05 | amaninacan | That makes even more sense. |
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22:16.46 | amaninacan | Since I've been out of VoIP for a while now, remind me again why someone would need any of those? |
22:18.40 | p3nguin | Some people use STUN when they have a very nasty NAT implementation that SIP/RTP do not work with. |
22:19.24 | p3nguin | I've never had an occasion to use port triggering. Ever. |
22:19.30 | amaninacan | Yeah. |
22:21.07 | amaninacan | Well, thanks p3nguin. That was exactly what I needed to hear. Now I can go tell these idiots to shut the fuck up and fix their shit, or lose my business. |
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22:46.15 | Dovid | hi all. my OpenSipS server sends INFO packets to see if asterisk is alive. asterisk gets the info packets but does not respond. any idea why asterisk would not do that? running asterisk 1.8.X |
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22:52.23 | CGMChris | One of my customers (the owner of his organization, with about 8 phones), he wants to dial an extension and have everybody's phone ring and everyone all jump into a conference. He's using a 4 line phone, so I'm guessing if this is possible it must us a function in the dial plan. Anyone ever tried to do this? |
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22:53.23 | WIMPy | core show application Page |
22:54.24 | CGMChris | WIMPy: That throws an error -> WARNING[17650]: app_meetme.c:800 build_conf: Unable to open pseudo device |
22:54.47 | WIMPy | Yes, you need dahdi for that. |
22:55.06 | pabelanger | CGMChris: dial all the extensions then redirect the channels to a local conference call when they answer |
22:55.14 | CGMChris | WIMPy: So, I need ztdummy since I don't have Zaptel hardware? |
22:55.40 | WIMPy | that would be dahdi today. |
22:56.02 | WIMPy | Or you do it manually from the dialplan and with ConfBrige. |
22:56.06 | CGMChris | pabelanger: Wouldn't that have to be linear...with one person having to answer before it could move on to the next Dial() request? Dial is not async right? |
22:56.33 | CGMChris | WIMPy: I'll go for the dahdi approach. Thanks. |
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22:58.02 | paulc | What's the highest number of accounts/line appearances you've seen on a softphone? Most seem to max out at 5 or 6 - anyone know of a softphone with more? |
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23:05.10 | pabelanger | CGMChris: you could work some dialplan magic to blast them out at the same time using local channels |
23:05.51 | CGMChris | I got it. Thanks for help, I just needed to do some yum magic to install "dahdi-tools", updated dahdi-firmware and dahdi-linux and reboot and its good to go. |
23:07.47 | p3nguin | paulc: 12 |
23:08.04 | paulc | p3nguin: That'd do nicely - what product? |
23:08.09 | p3nguin | paulc: Wait, you said soft phone. Disregard. |
23:08.13 | paulc | DOH! |
23:08.29 | paulc | (which phone with 12? I've got 9 on an Aastra something or other and it works great for what I want to do) |
23:08.39 | p3nguin | 5 or 6 seems about right for a soft phone. |
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23:10.51 | p3nguin | The SPA509G is a 12-line SIP phone. But with many phones, you can add expansion modules and have a crap-load more. |
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23:11.07 | CGMChris | Cisco has the SPA504G with 4 lines, and another 9 series with 8 lines, and they both support the 32 line attendant console...so if you need lines/appearances and can get away from the softphone into a physical device, that's the ultimaqte |
23:12.28 | [TK]D-Fender | attendant console != LINES |
23:12.33 | [TK]D-Fender | On Cisco... |
23:12.52 | [TK]D-Fender | Polycom IP601+ support 12 w/ and extra attendant module |
23:12.58 | paulc | Yeah.. no major problems using a physical device.. doing a bunch of testing with queues and stuff.. I like the Aastra cos it gives me 9 registrations |
23:13.16 | p3nguin | 12 > 9 |
23:13.59 | paulc | But $["12" > "9"] is false (as discovered today, randomly) |
23:14.32 | p3nguin | Always nice to compare number as text strings. |
23:15.59 | [TK]D-Fender | paulc: Without quotes it will parse as numeric properly |
23:16.01 | paulc | The clue was when the web service returned 6, it was tested against > "50" and returned true and I'm like "Hello WTF?" - then I realised what he'd done |
23:16.32 | paulc | [TK]D-Fender: Yup yup, did a quick bit of testing and realised "ah, it's properly strict" - so I'm fixing it all over the place now. |
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23:25.29 | p3nguin | paulc: Need more lines? http://www.digiumcards.com/images/digium_cards_images/sidecar_for_601.jpg |
23:26.27 | paulc | Haha lines to the max! Do the Polycom sidecars give you a registration per button then? I thought they were more BLF type functionality than registrations? |
23:29.37 | p3nguin | You can use them for all sorts of things, including registrations for multiple accounts. |
23:30.52 | Micc | in 1.8.8.1 when I park a call and it rings back, its ringing back the sip trunk that the caller is on, not the one that parked the call. Is this how it's supposed to work now? Its kind of hard deciphering how its supposed to work by reading features.conf |
23:31.06 | p3nguin | "configurable as a line registration, call appearance, or speed dial" |
23:31.52 | p3nguin | Parking should ring back to the device that parked the call. |
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23:35.10 | Micc | p3nguin, yeah it used to work in 1.6, but in 1.8 its ringing back to the trunk that the caller is on. |
23:35.28 | p3nguin | You know the drill: show me. |
23:35.48 | Micc | ok, but I gotta get some taco bell first. brb |
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