| 00:14.22 | *** join/#asterisk pcAngel (yoink@S0106602ad072bdf0.vc.shawcable.net) |
| 00:15.46 | pcAngel | Hey guys - I'm curious which queueing discipline I should be using on my SIP signalling & RTP rings - I'm using pfifo at the moment, and almost all examples online use SFQ |
| 00:16.48 | pcAngel | I figure that SFQ is a bad option because unless I'm getting thrashed or DOS'd by traffic matching the shaping policy for that ring, that I would want RTP & signalling going out in order, to reduce any delays |
| 00:21.25 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 00:25.26 | j4jackj | I don't know |
| 00:26.14 | j4jackj | But what I do know is that I have a conference service. I call it 'Dial 7', because before the conferencename, you dial 7. |
| 00:27.37 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.5.1 (2013/08/37), 10.12.3 (2013/08/27), 1.8.23.1 (2013/08/27), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 00:29.33 | *** join/#asterisk felipealmeida (~user@177.17.117.52) |
| 00:32.00 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-qqfbgfsmqzkdbejj) |
| 00:33.51 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 00:45.51 | *** join/#asterisk russum (~russum@ool-4a599ffe.dyn.optonline.net) |
| 00:47.32 | *** part/#asterisk russum (~russum@ool-4a599ffe.dyn.optonline.net) |
| 00:48.15 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 01:00.42 | j4jackj | I'm going to attempt to make a call to Google Voice via Asterisk, it seems they have moved Voice to Hangout so I'm going to see if this works. Should I dial 238004444444 (where 23 is the prefix I sue for Google Voice) in my softphone? |
| 01:01.02 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 01:03.22 | *** join/#asterisk petris (~petris@209.141.38.122) |
| 01:08.27 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 01:08.47 | j4jackj | Hello? |
| 01:10.08 | *** join/#asterisk Alpha64 (Alpha64@unaffiliated/alpha64) |
| 01:10.16 | j4jackj | [TK]D-Fender: does Google Voice still work with Asterisk? |
| 01:18.08 | andrewyager | Does anyone know how to surpress the stale nonce message from polycom handsets in the asterisk lgos? |
| 01:18.12 | andrewyager | *logs |
| 01:19.18 | j4jackj | I don't but I'm sure they're useful for something |
| 01:20.00 | andrewyager | In some situations they would be. Polycom's behaviour in this instance is actually probably wrong (much debate of whether they implement the standard correctly); but either way it's how they do it. |
| 01:21.32 | *** part/#asterisk Alpha64 (Alpha64@unaffiliated/alpha64) |
| 01:24.12 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:24.36 | j4jackj | Iksemel is needed for GV |
| 01:24.38 | j4jackj | Hmm |
| 01:24.49 | j4jackj | I'm going to recompile |
| 01:27.22 | *** join/#asterisk dukedukes (dukedukes@blk-89-228-4.eastlink.ca) |
| 01:37.45 | j4jackj | How long does an sterisk compile take? |
| 01:39.24 | WIMPy | Better ask how long vonfigure takes. |
| 01:39.32 | WIMPy | s/v/c/ |
| 01:41.02 | j4jackj | Why? Is asterisk compile done in a flash? |
| 01:41.04 | j4jackj | :D |
| 01:41.14 | Penguin | It usually takes a minute. |
| 01:41.42 | j4jackj | Heehee |
| 01:41.59 | j4jackj | That is what I mean by done in a flash... I thought it would be a fly by night compile. |
| 01:42.12 | WIMPy | For the first time compiling takes a minute. On upgrading configure and make install take almost all the time. |
| 01:43.04 | j4jackj | Configure is taking ages. |
| 01:51.40 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.247) |
| 01:53.01 | WIMPy | Maybe it's a good time for a little upgrade... |
| 01:55.00 | j4jackj | like? |
| 01:55.18 | j4jackj | Also, compile is taking more than 3 minutes... This is normal for my hardware. |
| 01:56.11 | WIMPy | What century technology do you have? |
| 01:56.20 | WIMPy | My PC is like 4 years old. |
| 01:57.12 | j4jackj | Mine is one year older, but it's a Celeron so I take long load times for granted |
| 01:57.35 | j4jackj | s/one year older/9-10 years old/ |
| 01:58.41 | j4jackj | WIMPy: so now you know, it works for what I use it for |
| 01:58.46 | WIMPy | I find compiling quite annoying on the P3. |
| 01:58.54 | j4jackj | Up to 5 minutes now and compile not finished |
| 01:59.11 | j4jackj | WIMPy: well this is a Pentium 4-style Celeron. |
| 01:59.44 | j4jackj | It's close to done though |
| 01:59.56 | j4jackj | It's well baked- |
| 01:59.57 | j4jackj | :D |
| 02:01.31 | j4jackj | Should I make as Asterisk Pie? It will be a pancake shaped like the asterisk logo. |
| 02:02.08 | WIMPy | Pi Pies! |
| 02:02.20 | j4jackj | I much prefer Asterisk Pies |
| 02:02.22 | j4jackj | :D |
| 02:03.01 | WIMPy | would be anxious they might be rotten inside. |
| 02:03.31 | j4jackj | But you spec your number of slices in Tau, like tau is the whole thing and tau/6 is a 6th |
| 02:03.54 | j4jackj | Why do you think they would be rotten? |
| 02:04.24 | WIMPy | That's what I associate with Asterisk. |
| 02:05.09 | j4jackj | Howso? How is Asterisk rotten? |
| 02:05.49 | WIMPy | To me it's shiney ornaments on a rotten foundation. |
| 02:06.03 | j4jackj | howso? |
| 02:06.13 | j4jackj | How is the foundation of Asterisk rotten? |
| 02:06.18 | WIMPy | Many long stories. |
| 02:06.26 | j4jackj | Pastebin them |
| 02:06.29 | j4jackj | Or PM me |
| 02:07.01 | WIMPy | You will find them when reading here or on jira. |
| 02:07.13 | j4jackj | huh? |
| 02:07.18 | j4jackj | You hate asterisk? |
| 02:07.36 | j4jackj | If so, WIMPy, gt*o of here. |
| 02:07.44 | WIMPy | No. It has its uses. And there are things it's really good at. |
| 02:08.05 | j4jackj | Thus, not a rotten foundation. |
| 02:08.15 | WIMPy | I think its unbeatable with IVRs and the new Confbridge is awesome. |
| 02:08.19 | j4jackj | It's useful for conference bridging... |
| 02:08.50 | j4jackj | And you don't have to use a softphone to use it (unlike Skype) |
| 02:08.58 | Penguin | You can have a really nice house, still built on a pile of sticks. |
| 02:09.10 | j4jackj | hehe |
| 02:09.11 | WIMPy | But the internals are often hair-raising. |
| 02:09.12 | Penguin | Bad foundation, nice stuff on top. |
| 02:09.59 | WIMPy | And I still don'e see it anywhere near being usefull as a replacement for a PBX. |
| 02:10.29 | Penguin | It makes a pretty decent PBX, actually. |
| 02:10.32 | andrewyager | WIMPy - it works quite well as a replacement PBX, but it's all around how you build the solution. |
| 02:10.38 | *** join/#asterisk italorossi (~italoross@187.61.168.117) |
| 02:10.51 | andrewyager | Asterisk is a great replacement PBX |
| 02:10.56 | WIMPy | Too many basics that don't work/exist. |
| 02:10.57 | andrewyager | but other things do much better jobs of other things |
| 02:11.13 | Penguin | Reinvent the wheel. |
| 02:11.41 | andrewyager | If you've got a favourite PBX vendor that does "X" or "Y" use them. Probably "could" be done in Asterisk, but I won't stand here and sell it! |
| 02:12.10 | andrewyager | We do PBX replacements all the time with Asterisk, but it's all around what you "add" to the box. |
| 02:12.11 | WIMPy | Earlier on I brought the example of using call forwarding. |
| 02:12.22 | WIMPy | And pickup from voicemail is another no go. |
| 02:12.35 | andrewyager | These things can be done; just require effort |
| 02:12.46 | andrewyager | Same as when you have a PBX handset not from the vendor of your PBX |
| 02:12.54 | WIMPy | And handling failed transfers is at least a little iffy. |
| 02:13.11 | WIMPy | No, they can't be done. |
| 02:13.12 | andrewyager | You get integration with other platforms (or, in the case of Broadsoft, they work really closely with the manufacturers to implement their functions and spec) but you loose |
| 02:13.16 | andrewyager | in other places |
| 02:13.20 | WIMPy | Not without programming your own app. |
| 02:13.33 | andrewyager | WIMPy: we've done call forwarding at PBX end several times. |
| 02:13.50 | andrewyager | Pickup from voicemail can't be done without writing your own voicemail app |
| 02:14.01 | WIMPy | I'm talking about real CF, not setting up another call. |
| 02:14.13 | andrewyager | Cool. As I said, not going to argue it! |
| 02:14.13 | andrewyager | :) |
| 02:14.33 | j4jackj | I can't do GV, because |
| 02:14.41 | j4jackj | I'm having GNUTLS issues |
| 02:14.53 | andrewyager | good luck then! They are fun… |
| 02:15.10 | Penguin | You need to havea iksemel and then rebuild gnutls, then rebuild asterisk. |
| 02:15.20 | WIMPy | And doing 3-way calling on the phone instead of a PBX isn't great at all. |
| 02:15.52 | j4jackj | I have to rebuld GnuTLS, Iksemel and then... what the asterisking hell? |
| 02:16.10 | andrewyager | Actually, it's really great according to the SIP standard, but that doesn't mean that it's good from an admin point of view. |
| 02:16.22 | andrewyager | Again, comes down to handset/PBX integration. You'll get different support from Avaya, Nortel, Cisco, etc |
| 02:16.26 | andrewyager | because they control the stack. |
| 02:16.49 | WIMPy | Administration of SIP phones is quite a big extra. |
| 02:16.52 | andrewyager | Don't buy a chicken and expect it to be a duck. |
| 02:17.03 | j4jackj | Asterisk is not the whole stack. |
| 02:17.09 | j4jackj | Which is a very good thing. |
| 02:17.11 | j4jackj | Heehe |
| 02:17.12 | andrewyager | But feel free to ask the question on whether or not you need a chicken or a duck. |
| 02:17.42 | WIMPy | Where's the beef? |
| 02:17.49 | andrewyager | Probably in the cow. |
| 02:18.00 | Penguin | If you cross a chicken and a duck, do you get a dicken or a chuck? |
| 02:18.07 | j4jackj | Chuck |
| 02:18.11 | j4jackj | Or ducken |
| 02:18.13 | WIMPy | I think I want a cow then. |
| 02:18.23 | j4jackj | dependng on whether the duck is the male or female |
| 02:18.39 | Penguin | Let's pretend they are both females. |
| 02:18.45 | andrewyager | Either way, it sounds tasty. |
| 02:18.46 | j4jackj | That's impossibl then |
| 02:19.31 | Penguin | You were thinking of natural reproduction. That's where you went wrong. |
| 02:19.37 | j4jackj | Hmm |
| 02:20.29 | Penguin | You've never heard of lesbians having children? It wasn't because they did it together by themselves. |
| 02:20.49 | j4jackj | Grr... |
| 02:21.04 | Penguin | You have to think outside the box. |
| 02:21.22 | j4jackj | If both chicken and duck are female, you would have an assploding asterisk. |
| 02:21.46 | j4jackj | With a lot of homosexual silliness and most importantly, no child. |
| 02:22.04 | andrewyager | Based on that definition, sounds like this might have been what happened. |
| 02:23.02 | Penguin | Two female PBXs thought it would be fun to have children? |
| 02:23.37 | WIMPy | wonders if Asterisk is female. |
| 02:23.47 | WIMPy | Would make sense, I guess. |
| 02:24.05 | Penguin | or an emo male. |
| 02:28.22 | j4jackj | Asterisk is a male, whose wife des all the talking (she likes doing that) |
| 02:28.23 | j4jackj | :D |
| 02:30.06 | *** join/#asterisk iamamoron (~Miranda@aa20120301773d781390.userreverse.dion.ne.jp) |
| 02:30.38 | iamamoron | hi there i have an existing analog PBX and i want to integrate asterisk on it would it possible? |
| 02:31.37 | Penguin | Are you a moron? |
| 02:31.52 | Penguin | (I initially thought it said you were a mormon.) |
| 02:36.00 | igcewieling | he could be a mormon |
| 02:36.44 | igcewieling | iamamoron: you can do some limited integration, but generally it is actually more work and almost as expensive as replacing the pbx with Asterisk |
| 02:39.08 | iamamoron | that is why i am here asking because i dont have dam idea |
| 02:39.20 | iamamoron | so it cannot be done? |
| 02:39.34 | iamamoron | http://www.digium.com/en/products/voip-gateways/g200 |
| 02:39.39 | iamamoron | that one cannot work? |
| 02:42.39 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-56-12.rn.hr.cox.net) |
| 02:43.11 | slav3_kitten | i'm a bit confused.... |
| 02:43.19 | slav3_kitten | what are you traying to do iamamoron ? |
| 02:43.40 | iamamoron | i have an existing pbbx panasonic |
| 02:43.47 | iamamoron | pbx |
| 02:44.11 | slav3_kitten | and it's on a T1? |
| 02:44.20 | iamamoron | then i wanted to use IP phone in order for me to expand immediate as long as there is lan cable i can have phone also |
| 02:44.55 | iamamoron | i needed the two technologies to work togethere |
| 02:45.44 | slav3_kitten | so you need a old school pbx to also do voip pbx? |
| 02:45.59 | iamamoron | yes |
| 02:46.07 | iamamoron | i want them to work together |
| 02:46.14 | iamamoron | new implementation is Ip based |
| 02:46.19 | iamamoron | old one as is |
| 02:46.35 | slav3_kitten | that's a pain in the ass... i think you'd end up creating a very hard to maintain system doing it that way |
| 02:46.41 | slav3_kitten | vs replacing everything with voip |
| 02:47.06 | slav3_kitten | you could do ip phones to a T1 |
| 02:47.36 | Penguin | Disconnect your phone lines from the analog PBX. Connect the lines to the proper interface card in an asterisk-based PBX. |
| 02:48.10 | slav3_kitten | exactly Penguin |
| 02:48.29 | iamamoron | but i needed to preserve the old one |
| 02:48.32 | iamamoron | that is many |
| 02:48.47 | slav3_kitten | how many phones are on the old system? |
| 02:48.55 | iamamoron | 30 |
| 02:49.11 | Penguin | Asterisk works well with 30 phones. |
| 02:49.11 | iamamoron | properly wired |
| 02:49.20 | Penguin | 30 IP phones |
| 02:49.25 | iamamoron | and i cannot afford to have downtime |
| 02:50.07 | slav3_kitten | do you run 24/7? |
| 02:50.21 | iamamoron | yeap |
| 02:50.24 | iamamoron | that is a warehouse |
| 02:50.29 | iamamoron | they operate 24hours |
| 02:50.42 | slav3_kitten | schedule downtime |
| 02:50.48 | iamamoron | that is why if possible i can integrate |
| 02:51.11 | iamamoron | how about the t1 you mena? |
| 02:51.15 | iamamoron | mean? |
| 02:51.19 | slav3_kitten | plan your project, figure your downtime, replace it right the first time instead of ending up with downtime in the future from trying to mesh the two |
| 02:51.22 | iamamoron | ip phones on T1 |
| 02:52.04 | slav3_kitten | you could have your T1 24 line hook into an asterisk server that is the new pbx and serves all of the phones |
| 02:52.15 | slav3_kitten | if i'm not mistaken |
| 02:52.31 | Penguin | Yep. Just need the right interface for the new asterisk box. |
| 02:53.12 | Penguin | Digium should have a T1 card that will work in the computer hardware that you get for the new PBX. |
| 02:53.36 | iamamoron | around how much is that? |
| 02:53.39 | slav3_kitten | and a fxs card to handle all the analog lines you need to keep supporting |
| 02:53.48 | iamamoron | how can i connect it from old pbx? |
| 02:54.22 | deegen | That is, unless your old pbs packs it in before all your phones are in. |
| 02:54.40 | Penguin | Sounds like he needs to get started. |
| 02:54.41 | iamamoron | i dont understand |
| 02:54.48 | slav3_kitten | no offense or anything here... but if you don't know how to do this, maybe you should consult with a 3rd party that knows phone systems |
| 02:55.11 | iamamoron | who wants to be consultant? |
| 02:55.23 | iamamoron | how much? |
| 02:55.34 | slav3_kitten | not i, i just run a small 6 phone home network |
| 02:55.39 | Penguin | We're telling you how to handle it. For free. |
| 02:55.54 | *** join/#asterisk andrewyager (~andrewyag@syd02s26-fw01.thecore.net.au) |
| 02:56.06 | iamamoron | ok Penguin am glad to hear that |
| 02:56.07 | iamamoron | but |
| 02:56.15 | iamamoron | i wanted to know how to connect that |
| 02:56.19 | slav3_kitten | iamamoron, might want to get the book an read it cover to cover |
| 02:56.24 | Penguin | ~book |
| 02:56.24 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 02:56.42 | Penguin | What type of phone lines do you have on the old PBX? |
| 02:57.08 | Penguin | (2157.06) Irssi uptime: 462d 22h 54m 52s |
| 02:57.10 | Penguin | hmm |
| 02:57.13 | Penguin | That's a while. |
| 02:57.20 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 02:57.30 | iamamoron | analog phones |
| 02:57.56 | iamamoron | connected to panasonic crap |
| 02:58.01 | slav3_kitten | right but is it say 24 individual lines, or is it a single T1/E1 line coming in that provides your 24-30 lines |
| 02:58.09 | Penguin | Lines. Lines, not phones. |
| 02:58.31 | Penguin | Lines come from the telco. |
| 02:58.41 | iamamoron | PSTN |
| 02:58.55 | iamamoron | it is still PSTN |
| 02:59.03 | slav3_kitten | we get that it's PSTN, there are multiple types of PSTN |
| 02:59.06 | Penguin | What KIND OF LINES? |
| 02:59.28 | slav3_kitten | like T1/E1, standard 2 wires per line coming in from the demarcation point |
| 02:59.32 | Penguin | Do you have 50 pairs? |
| 02:59.47 | iamamoron | wait ill check |
| 02:59.55 | slav3_kitten | take a picture if you can |
| 03:01.15 | igcewieling | iamamoron: First you need to learn Asterisk. Once you lean Asterisk then you will understand how to connect them. Start by reading the Asterisk book. |
| 03:01.17 | igcewieling | !book |
| 03:01.24 | iamamoron | from telco it is a copper wire terminated by a telephone box and from there the pbx is connected |
| 03:01.44 | Penguin | One pair? |
| 03:01.45 | iamamoron | thats the set up |
| 03:01.55 | iamamoron | i have 3 lines |
| 03:01.59 | Penguin | Three pairs? |
| 03:02.00 | iamamoron | 3 telephone numbers |
| 03:02.10 | igcewieling | sounds like analog to me |
| 03:02.18 | iamamoron | yes it is still analog |
| 03:02.32 | Penguin | So you need a card with three FXO ports. |
| 03:02.51 | iamamoron | ahh you mean will disconnect the current pbx now? |
| 03:02.53 | Penguin | or four if you want to expand later. |
| 03:03.00 | slav3_kitten | iamamoron, get a picture of it |
| 03:03.04 | Penguin | You'll throw the current PBX right into the bin. |
| 03:03.14 | igcewieling | iamamoron: expect plenty of downtime no matter what you do |
| 03:03.16 | iamamoron | oach |
| 03:03.21 | iamamoron | cant afford that |
| 03:03.32 | Penguin | Then sell it on ebay. |
| 03:03.35 | igcewieling | you most common issue will be stuck calls between Asterisk and the analog PBX. |
| 03:03.42 | slav3_kitten | iamamoron, plan your downtime. do the work when it's least being used |
| 03:04.30 | iamamoron | thanks for the advise |
| 03:04.31 | igcewieling | You might be able to minimize the issue by using FXS cards in Asterisk connected to the PBX CO ports, but you will have issues if you connect Asterisk fxo ports to the PBX FXS ports (if it even has analog FXS ports) |
| 03:04.32 | slav3_kitten | downtime is unavoidable when upgrading core systems, but the inconvenience can be mitigated by telling everyone when you'll be down, when you plan on being up, and working when it's least being utilized |
| 03:04.40 | iamamoron | i need to think more on this one |
| 03:04.54 | igcewieling | iamamoron: and read the Asterisk book. PBXs are COMPLICATED |
| 03:04.54 | j4jackj | I think I will use YATE for GVoice support |
| 03:11.36 | Penguin | Weird. |
| 03:14.27 | j4jackj | I'm trying to find out how to change my Google Voice account type... |
| 03:14.36 | Penguin | From what to what? |
| 03:15.52 | j4jackj | From whatever it is now to 'With a google number' |
| 03:16.25 | Penguin | You didn't choose the phone number when you got the account? |
| 03:17.06 | j4jackj | I don't think so no |
| 03:17.19 | j4jackj | AFAIK |
| 03:17.36 | Penguin | I don't remember where you go to choose your number. |
| 03:17.44 | j4jackj | Nor do I |
| 03:17.49 | Penguin | It's been years since I did mine. |
| 03:17.54 | j4jackj | Maybe it's an on signup only option |
| 03:18.07 | Penguin | You can create the account without choosing the number. |
| 03:18.28 | j4jackj | I think I did that |
| 03:18.33 | Penguin | But you can't really do anything else until you get the number selected. At least that's how it was. |
| 03:18.52 | *** join/#asterisk russum (~russum@ool-4a599ffe.dyn.optonline.net) |
| 03:18.59 | j4jackj | Where would you go to un-sign-up from GV without deleting your whole GOOG acct? |
| 03:19.17 | Penguin | I don't think you can do that on your own. You'd have to ask google to do it. |
| 03:19.33 | phix | I don't know what any of those abreviations / acronyms mean |
| 03:19.50 | Penguin | Which ones, phix? |
| 03:21.01 | phix | GV, GOOG |
| 03:21.16 | phix | Google related? |
| 03:21.18 | j4jackj | GV = Google Voice, GOOG=Google's stock ticker |
| 03:21.21 | phix | ah |
| 03:22.22 | phix | ummmm With a google account you have access to all of the google services, including drive, play, docs, reader, google+ etc etc., if you don't want to use one of them then just don |
| 03:22.26 | phix | t use it :) |
| 03:23.08 | j4jackj | I'm trying to recreate the accountwith different settings... |
| 03:23.17 | Penguin | There aren't settings. |
| 03:23.22 | j4jackj | I know... |
| 03:23.31 | j4jackj | I'm now trying to find out what happened... |
| 03:23.41 | Penguin | You either initiated the voice account or you didn't. |
| 03:23.46 | Penguin | If you did, you should be able to get a phone number. |
| 03:23.58 | j4jackj | I did start the acct |
| 03:24.03 | Penguin | Once you have the number, then you can set different call settings and voiec mail stuff, etc. |
| 03:24.17 | j4jackj | Now I don't know what happened. can't set anything except billing and timezone. |
| 03:25.14 | j4jackj | I can only set billing and timezone. That sux. |
| 03:26.56 | igcewieling | phix: all of those abbreviations actually mean "I'm too cheap to sign up with an ITSP" |
| 03:28.00 | j4jackj | igcewieling: well I am :D |
| 03:28.53 | igcewieling | you know how much time I spend "managing" my ITSP account per month? none. |
| 03:28.57 | phix | igcewieling: heh |
| 03:29.24 | Penguin | I spend just almost that much time on mine. |
| 03:29.42 | igcewieling | if you just want peer to peer calls, you don't need Google services, if you want to be able to call to/from the PSTN get an account with an ITSP. |
| 03:29.46 | j4jackj | Are the betamax providers useful...? |
| 03:30.04 | igcewieling | j4jackj: you speak words, but they don't always make sense. |
| 03:30.12 | j4jackj | igcewieling: yup |
| 03:30.20 | igcewieling | ~itsp |
| 03:30.20 | 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. |
| 03:30.38 | j4jackj | ~itsplist-ca |
| 03:30.38 | infobot | rumour has it, itsplist-ca is Here are some popular Canadian ITSPs: http://www.les.net , http://www.babytel.ca , http://www.voip.ms, http://unlimitel.ca |
| 03:30.44 | Penguin | I'm interested in this betamax provider concept. |
| 03:30.56 | j4jackj | Ever heard of Dellmont Sarl? |
| 03:31.06 | j4jackj | They run all the betamax clone-s |
| 03:31.17 | j4jackj | They're a telephone provider... |
| 03:31.43 | *** part/#asterisk russum (~russum@ool-4a599ffe.dyn.optonline.net) |
| 03:34.39 | j4jackj | look at things like poivy.com nd voipcheap.com and you'll see what I mean. All the same website. |
| 03:35.18 | Penguin | SIgn up with VoIP.ms, put some money on the account, enjoy. |
| 03:35.32 | carrar | SOUNDS ILLEGAL |
| 03:35.48 | j4jackj | carrar: it is not |
| 03:36.56 | carrar | soldering iron and sounds cards and you got yourself a FINE ATA |
| 03:36.59 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 03:36.59 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 03:37.05 | carrar | Thats all one needs to know |
| 03:38.34 | Penguin | ~savemoney |
| 03:38.34 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 03:38.53 | carrar | heh |
| 03:41.10 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-awyytzocxiudsudn) |
| 03:48.37 | *** join/#asterisk tyman (~tyman@173-12-219-184-Fresno.hfc.comcastbusiness.net) |
| 03:51.23 | igcewieling | 5.1 Surround Sound Telephony |
| 03:51.42 | Penguin | 5.1 is so 2000. |
| 03:52.03 | igcewieling | I still use 2.0 surround sound. 8-| |
| 03:52.25 | Penguin | I have 2.1, so I feel your pain. |
| 03:52.39 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.72) |
| 03:52.48 | Nugget | that's epic |
| 03:54.27 | Nugget | switches carrar from CrO to METAL |
| 04:02.24 | *** join/#asterisk russum (~russum@ool-4a599ffe.dyn.optonline.net) |
| 04:03.23 | j4jackj | It's OK, but I was hoping to use my Intel 537/537EP/538 modem as a Zap card. Oh well. :D |
| 04:04.53 | j4jackj | :D |
| 04:06.15 | Nugget | is re-reading the Gremlin logs |
| 04:06.18 | j4jackj | :( No noe is answering.... |
| 04:06.22 | Nugget | I forgot how good this was |
| 04:07.59 | *** join/#asterisk evilman_work (~evilman@87.244.6.228) |
| 04:08.18 | igcewieling | The Intel card thing was only funny for the first 10 years. |
| 04:08.55 | Nugget | Hi everyone, I need help. I'm trying to run trixbox with realtime patches on an eMachines server I found in the storage closet. I've got 40 users with a mixture of grandstream phones and x-lite (unregistered). I'm using four clone x100p cards I bought off ebay and I compiled a pirated version of the g729 codec. Can you help me set up fax over sip? |
| 04:08.56 | j4jackj | heh |
| 04:09.14 | j4jackj | Nugget: that will get you banned |
| 04:09.30 | j4jackj | Use G722 or G711. |
| 04:09.37 | j4jackj | Or GSM |
| 04:09.59 | j4jackj | igcewieling: is it even possible to use an Intel card with asterisk? |
| 04:10.57 | *** join/#asterisk CeBe1 (~CeBe@port-92-206-53-172.dynamic.qsc.de) |
| 04:11.35 | igcewieling | j4jackj: not any made in since 2003 or so. |
| 04:11.48 | j4jackj | I mean a 537 |
| 04:11.52 | j4jackj | Or a 538 |
| 04:11.57 | j4jackj | IDK which one I have |
| 04:12.50 | igcewieling | There were two modems which could be used as a zap card. they were discontinued around 2003 or so. |
| 04:13.57 | j4jackj | igcewieling: Well mine is a winmodem of some kind. |
| 04:14.09 | j4jackj | I don't have the model number... |
| 04:14.10 | igcewieling | j4jackj: was it made before 2002? |
| 04:14.21 | j4jackj | IDk |
| 04:14.24 | j4jackj | Let me get it |
| 04:14.29 | igcewieling | then throw it in the trash |
| 04:14.38 | igcewieling | Regardless you are on your own |
| 04:16.09 | j4jackj | I heard that 537 cards work. |
| 04:22.06 | j4jackj | This is going to get rather interesting |
| 04:23.37 | j4jackj | I'm trying to hotplug the modem I think is a 536/537 and the computer can't see the modem. |
| 04:24.15 | ChannelZ | That isn't what "plug and play" means. |
| 04:25.33 | j4jackj | I get that much |
| 04:25.48 | j4jackj | Must I now reboot the computer? |
| 04:26.05 | j4jackj | Or should I lspci and see what happens? |
| 04:26.17 | ChannelZ | did you seriously jam an expansion card into a running machine? |
| 04:26.56 | j4jackj | yup |
| 04:27.07 | j4jackj | did I just b0rk the expansion |
| 04:28.05 | j4jackj | Do I now have a piece of inedible chip toash? |
| 04:28.49 | j4jackj | BTW it's PCI |
| 04:29.50 | j4jackj | ChannelZ: ? |
| 04:33.51 | j4jackj | Anyone? |
| 04:38.15 | j4jackj | By hotplugging a PCI expansion card into my computer, what did I just b0rk? |
| 04:53.06 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.45) |
| 04:54.31 | Penguin | I would power off and power on. |
| 04:54.46 | Penguin | You could have zapped it, but power down and up again to see if it works. |
| 04:59.56 | j4jackj | I'm not going to. |
| 05:00.18 | j4jackj | For all I know, 16 people could be using my computer for conference alone. |
| 05:04.08 | Penguin | I can't imagine you could get 16 people interested in calling you at the same time. |
| 05:04.40 | j4jackj | heh |
| 05:05.21 | Penguin | I can't imagine you could get 16 different people interested in calling you at different times. |
| 05:05.27 | Penguin | cumulatively. |
| 05:06.10 | Penguin | Oops, wife says it's bed time. Better stop giving you crap and go. |
| 05:08.30 | ChannelZ | and also, "for all you know" is pretty easy to find out. |
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| 06:50.26 | *** join/#asterisk bombev (~User@PPPoE-Static-40-132.UnicsBG.Net) |
| 06:51.54 | bombev | good morning |
| 06:52.04 | *** join/#asterisk izbushka (~izbushka_@193.23.225.11) |
| 06:54.56 | j4jackj | hi all |
| 06:55.20 | bombev | I am setting variable in regular context line 8, and I am not able to retrieve this variable in macro context line 19 - http://pastebin.com/Z8FxwMs6 |
| 06:56.05 | bombev | here you go the output http://pastebin.com/zYxgyKX3 |
| 06:56.22 | *** join/#asterisk jsjc (~Adium@221.Red-83-41-76.dynamicIP.rima-tde.net) |
| 06:56.41 | kaldemar | bombev: feed it as an argument to the macro. |
| 06:58.03 | j4jackj | kaldemar: i think you nailed it |
| 06:58.34 | j4jackj | kaldemar: altho for those with dynamic IPs STUN IS needed. |
| 06:58.55 | kaldemar | bombev: also, M(dendax-registration,s,1) should be just M(dendax-registration). all the option wants is a macro name. |
| 06:59.11 | bombev | aha |
| 06:59.11 | kaldemar | j4jackj: why? |
| 06:59.24 | bombev | got it |
| 06:59.28 | kaldemar | bombev: see what "core show application Dial" says about the syntax. |
| 06:59.57 | j4jackj | kaldemar: because otherwise you have to change the IP address inside your SIP client et al every day orso |
| 06:59.58 | bombev | but about retrieving variable whitin Macro context what should I fix in my dialplan |
| 07:00.20 | kaldemar | j4jackj: no you don't. |
| 07:00.40 | j4jackj | kaldemar: not withSTUN or TURN |
| 07:01.29 | kaldemar | bombev: the macro is executed on the called channel. that is a different channel from the calling channel, which has all the variables you defined. try func MASTER_CHANNEL |
| 07:02.08 | kaldemar | j4jackj: or using DNS. you're just coming up with needs/requirements that don't exist. |
| 07:02.29 | kaldemar | j4jackj: or by configuring asterisk properly. |
| 07:02.30 | j4jackj | kaldemar: through dynamic dns. that's flimsy though |
| 07:03.00 | kaldemar | the client does not even need to know its public address if you use asterisk. |
| 07:04.07 | bombev | so kaldemar the only solution is MASTER_CHANNEL |
| 07:04.11 | j4jackj | kaldemar: it des if both are behind a nat |
| 07:04.16 | bombev | what about GLOBAL Variable |
| 07:05.02 | kaldemar | bombev: concurrency will screw that up. |
| 07:05.15 | kaldemar | j4jackj: wrong again. |
| 07:05.22 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:79ab:6405:8635:b2a4) |
| 07:05.49 | j4jackj | kaldemar: you're a troll. we need to talk about this. call 71@99.199.11.127 and I'll be on the line shortly |
| 07:05.56 | *** join/#asterisk jhlavacek (~jirka@78.208.220.3) |
| 07:06.03 | bombev | kaldemar what about this http://www.voip-info.org/wiki/view/Asterisk+func+shared |
| 07:06.32 | kaldemar | bombev: that's another method. you'd need to pass the channel name as an argument. did you try MASTER_CHANNEL? |
| 07:07.03 | kaldemar | j4jackj: we do not need to talk about this. you need to do some reading and stop trying to waste my time. |
| 07:07.09 | bombev | not yet I am trying to find information about MASTER_CHANNEL because I have never used so far |
| 07:07.24 | kaldemar | bombev: "core show function MASTER_CHANNEL" |
| 07:07.27 | j4jackj | kaldemar: what reading do you suggest? because all I have known is 'get static or use STUN' |
| 07:07.39 | kaldemar | ~sipnat |
| 07:07.39 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 07:07.47 | *** join/#asterisk felimwhiteley_ (~quassel@89.101.203.26) |
| 07:08.14 | kaldemar | ~book |
| 07:08.14 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 07:08.18 | j4jackj | ... |
| 07:08.19 | kaldemar | some networking 101. |
| 07:08.27 | j4jackj | I don't read books that cost more than about $.01 |
| 07:08.54 | j4jackj | I don't need tounderstand the electrical interface of Ethernet to use it now do I? |
| 07:08.59 | kaldemar | you can read it free online, if you bother to read what infobot says. |
| 07:09.18 | j4jackj | Nor do I need to understand how PS/2 works to use a PS/2 keyboard |
| 07:09.46 | j4jackj | kaldemar: also, I don't read anything that will hurt my eyes. good night. |
| 07:20.15 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:22.40 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:22.58 | ChannelZ | lalalalala I can't hear you lalalalala |
| 07:23.56 | *** join/#asterisk giucio (~giucio@85.118.242.195) |
| 07:32.49 | giucio | Hi, in my company we have an Active Directory LDAP server which handles user authentication for everybody. I'm trying to configure asterisk so that users can log in using their LDAP username and password. The documentation (https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver) seems to imply that the hashed password can be retrieved from the LDAP server. Unfortunately With Microsoft Active Directory this cannot be done, you can only bind with yo |
| 07:33.55 | giucio | I'm using Asterisk 11.4.0 |
| 07:37.29 | Gugge | Damn, i guess this is not good. :) |
| 07:37.36 | Gugge | asterisk[14854] trap divide error ip:7ffd8b692479 sp:7ffd5f0791e0 error:0 in res_musiconhold.so[7ffd8b68d000+b000] |
| 07:53.10 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 08:02.34 | *** join/#asterisk j4jackj (jack@j4jackj-1-pt.tunnel.tserv21.tor1.ipv6.he.net) |
| 08:06.17 | wdoekes | Gugge: interesting |
| 08:06.26 | wdoekes | https://issues.asterisk.org/jira/browse/ASTERISK-21775 |
| 08:07.47 | wdoekes | and you can run addr2line on the ip:-number to get more info |
| 08:08.16 | wdoekes | $ echo 7ffd8b692479 | addr2line -Cfe /path/to/res_musiconhold.so |
| 08:08.29 | *** join/#asterisk g_r_eek (~g_r_eek@ppp-94-69-13-52.home.otenet.gr) |
| 08:18.10 | giucio | Is there a way to authenticate sip peers calling an external script? |
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| 08:57.56 | Gugge | wdoekes: that just outputs "??\n??:0" :) |
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| 09:05.32 | wdoekes | Gugge: too bad.. sometimes that prints more info |
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| 09:11.52 | Gugge | wdoekes: yep, ive used that before when the crash was in asterisk |
| 09:12.08 | Gugge | but seems it does not work with res_musiconhold.so :) |
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| 09:37.00 | bombev | kaldemar I did it, I just set global variable :) that works for me exten => _XX.,n,Set(GLOBAL(CASENUMBER)=${caseNumber}) |
| 09:37.12 | bombev | without using MACRO_CHANNEL function |
| 09:38.39 | kaldemar | i didn't say MACRO_CHANNEL but MASTER_CHANNEL. |
| 09:39.48 | bombev | yes MASTER_CHANNEL sorry |
| 09:39.55 | kaldemar | think about using a global variable and what happens if two calls use the extension simultaneously. |
| 09:41.42 | bombev | kaldemar what do you mean? |
| 09:42.46 | bombev | kaldemar but this MASTER_CHANNEL is ot available in Asterisk 1.8 :( so I cant use it |
| 09:52.35 | *** join/#asterisk vlad_starkov (~vlad_star@194.154.71.230) |
| 10:12.09 | kaldemar | bombev: if you set some value to a variable, and a another call comes in and sets it before the first value is used, you have unexpected behavior. |
| 10:12.43 | bombev | oh yes you are right |
| 10:13.07 | kaldemar | is there a reason you can't use arguments for the macro? |
| 10:13.52 | bombev | give me an example how to use arguments |
| 10:14.14 | kaldemar | "core show application Dial" |
| 10:14.25 | kaldemar | look at the M() option syntax. |
| 10:15.50 | kaldemar | in the macro, the arguments are in variables ARG1 .. ARGn. |
| 10:17.46 | bombev | yes kaldemar this is what I use now |
| 10:17.47 | bombev | exten => _XX.,n,Dial(SIP/160,40,mM(dendax-registration,s,1)) |
| 10:18.20 | kaldemar | i already said something about that... |
| 10:18.23 | bombev | this is in regular context exten => _XX.,n,Set(GLOBAL(CASENUMBER)=${caseNumber}) |
| 10:18.30 | bombev | this is in macro context exten => s,n,SayDigits(${CASENUMBER}) |
| 10:18.51 | bombev | s,1 is removed this is the old code |
| 10:19.15 | kaldemar | well don't paste invalid stuff then. |
| 10:19.47 | bombev | <PROTECTED> |
| 10:20.16 | bombev | ok this is the whole context http://pastebin.com/E4arxt6K |
| 10:20.34 | bombev | if I am not suppose to use Set(GLOBAL()) |
| 10:20.43 | bombev | how I can make it other way |
| 10:21.17 | kaldemar | which part of using arguments for the macro do you not understand? |
| 10:21.19 | wdoekes | Gugge: if you can attach relevant info to the ticket I mentioned, that'd be nice. e.g. your musiconhold config (and whether or not you're using realtime) |
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| 10:46.48 | izbushka | hi |
| 10:46.55 | izbushka | I want to allow users to set the call forwarding. Point me to the right direction please |
| 10:50.55 | izbushka | it's not a problem to set forwarding number. I'm looking for the convenient way to make redirects in Dialplan. Is there anything except checking redirects manually before Dial()? What to do with queues? |
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| 11:33.22 | bombev | kaldemar well can you give me an example |
| 11:33.33 | bombev | with Argument whitin macro context |
| 11:37.57 | *** join/#asterisk SeanN (~Sean_Nien@cpc1-hers1-0-0-cust582.6-3.cable.virginmedia.com) |
| 11:38.52 | SeanN | hi everyone |
| 11:38.55 | SeanN | I have an issue with originating calls via Asterisk Call Manager, is there anyone here who I could discuss my config with and get some guiance on where I may be going wrong? |
| 11:40.05 | WIMPy | You mean AMI? |
| 11:50.00 | *** join/#asterisk PLMg (PLMg@78.96.151.225) |
| 11:51.48 | SeanN | Hi WIMPy |
| 11:52.09 | SeanN | Asterisk Call Manager/1.3 |
| 11:52.20 | SeanN | via telnet on TCP:5038 |
| 11:52.39 | SeanN | I'm using the current version of Elastix |
| 11:52.46 | WIMPy | Cisco has a Call Manager. |
| 11:53.19 | WIMPy | That thing is called the Asterisk Management Interface, or AMI for short. |
| 11:53.28 | Greenlight | That's how the AMI identifies itself when you connect |
| 11:53.35 | SeanN | ok thanks |
| 11:54.02 | WIMPy | Does it? /me never noticed. |
| 11:54.07 | SeanN | the issue I have is calls says queued, but nothing happens |
| 11:54.14 | WIMPy | evil |
| 11:54.50 | WIMPy | Then you're probably trying to call something that isn't reachable. |
| 11:55.08 | SeanN | Response: Success |
| 11:55.08 | SeanN | Message: Originate successfully queued |
| 11:55.32 | SeanN | then I get: |
| 11:55.32 | SeanN | Event: OriginateResponse |
| 11:55.32 | SeanN | Privilege: call,all |
| 11:55.32 | SeanN | Response: Failure |
| 11:55.32 | SeanN | Channel: SIP/7868@from-internal |
| 11:55.33 | SeanN | Context: from-internal |
| 11:55.33 | SeanN | Exten: 907........... |
| 11:55.34 | WIMPy | Either use debug for the channel you use or use a local channel and dialplan to get debug that way. |
| 11:55.34 | SeanN | Reason: 0 |
| 11:55.34 | SeanN | Uniqueid: <null> |
| 11:55.35 | SeanN | CallerIDNum: 7868 |
| 11:55.35 | SeanN | CallerIDName: <unknown> |
| 11:55.38 | WIMPy | ~pb |
| 11:55.39 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 11:55.48 | SeanN | fair enough, sorry |
| 11:56.05 | WIMPy | That doesn't look like a valid channel. |
| 11:56.11 | SeanN | in the Asterisk CLI, verbose set to 3, I don't see anything |
| 11:56.38 | SeanN | what I'm trying to do is connect a SIP extension and an external number (a mobile) |
| 11:56.43 | WIMPy | Unless you have a device called from-internal. |
| 11:56.49 | SeanN | right |
| 11:57.42 | SeanN | changed channel to Channel: SIP/7868, something happened, half a ring then a hang up |
| 11:57.46 | SeanN | but that's something! |
| 11:57.56 | SeanN | I'll test a little more and come back |
| 11:59.02 | SeanN | Timeout: 1000 -> Timeout: 10000 |
| 11:59.03 | SeanN | all working |
| 11:59.07 | SeanN | thanks very much |
| 12:01.06 | bombev | WIMPy could you help me :) for one thing |
| 12:01.50 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 12:02.11 | *** join/#asterisk Maliuta (nikolai@donetsk.lusan.id.au) |
| 12:02.18 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 12:04.45 | kaldemar | bombev: i already did earlier. |
| 12:05.24 | bombev | kaldemar please give me an example for this argument in the macro context |
| 12:05.41 | bombev | because I dont get it somehow |
| 12:07.11 | kaldemar | what did you try? |
| 12:07.28 | cusco | 2013-08-28 13:02:53] WARNING[9237] app_queue.c: Failed pausing realtime queue member tphes-nps:Local/701@agents |
| 12:07.31 | cusco | [2013-08-28 13:02:53] VERBOSE[9237] app_queue.c: -- Failed to pause Queue Member Local/701@agents in queue tphes-nps! |
| 12:07.34 | cusco | why the hell ??? |
| 12:10.45 | Gugge | wdoekes: i attached a little info to the ticket |
| 12:11.15 | Gugge | cusco: does it always fail, or just sometimes? |
| 12:11.26 | bombev | kaldemar exten => _XX.,n,Set(CASENUMBER=${caseNumber}) in the regular context |
| 12:11.28 | bombev | and |
| 12:11.41 | bombev | exten => s,n,SayDigits(${ARG1}) |
| 12:11.44 | bombev | in the macro context |
| 12:12.05 | kaldemar | bombev: did you read what "core show application Dial" says about the M() option? |
| 12:12.08 | *** join/#asterisk Rumbles (~Rumbles@host81-133-199-16.in-addr.btopenworld.com) |
| 12:12.19 | bombev | yes |
| 12:12.31 | kaldemar | bombev: you have to pass an argument to the macro or ARG1 will be empty. |
| 12:12.57 | kaldemar | bombev: what did the documentation say about the syntax? |
| 12:13.07 | bombev | <PROTECTED> |
| 12:13.29 | kaldemar | the macro won't magically know that you want a variable called CASENUMBER to be an argument for the macro. |
| 12:17.25 | bombev | exten => _XX.,n,Dial(SIP/160,40,mM(dendax-registration^CASENUMBER)) |
| 12:17.30 | bombev | ? |
| 12:17.45 | wdoekes | thanks Gugge, and cachertclasses=yes ? |
| 12:18.17 | WIMPy | bombev: Name vs value. |
| 12:18.46 | Gugge | wdoekes: yes, forgot to write that. cachertclasses=yes is set |
| 12:19.00 | kaldemar | bombev: what does that set ARG1 to? |
| 12:20.55 | bombev | kaldemar WIMPy not it is ok exten => _XX.,n,Dial(SIP/160,40,mM(dendax-registration,${CASENUMBER})) |
| 12:20.56 | bombev | :) |
| 12:22.33 | *** join/#asterisk Vann (~manny@c-76-97-51-58.hsd1.ga.comcast.net) |
| 12:23.44 | *** join/#asterisk dgeary2 (~debian@pa49-187-94-200.pa.nsw.optusnet.com.au) |
| 12:24.00 | bombev | thank you guys |
| 12:26.24 | *** join/#asterisk dongola7 (~dongola7@50-197-3-153-static.hfc.comcastbusiness.net) |
| 12:26.34 | bombev | so kaldemar this way is better to use Macro and Arguments instead of to set GLOBAL variable ? |
| 12:27.09 | *** join/#asterisk vlad_starkov (~vlad_star@213.33.164.42) |
| 12:37.33 | *** join/#asterisk wmchris (~wmchris@ip-88-153-165-111.unitymediagroup.de) |
| 12:39.06 | wmchris | hi. need a lil hint. i'm using chan_dongle to integrate my 3G Stick into asterisk and using SIP to connect a phone. works w/o any problem, except the CLIP Support. It displays dongle0 as caller, instead of the caller id. how can i tell the system to submit the correct caller id to a sip client? |
| 12:44.15 | *** join/#asterisk felipealmeida (~user@177.17.117.52) |
| 12:46.30 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
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| 12:53.52 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:54.05 | *** join/#asterisk wmchris (~wmchris@ip-88-153-165-111.unitymediagroup.de) |
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| 12:56.03 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 12:57.45 | protocoldoug | So, security announcement last night, I'm going to update my systems, but... I've been running asterisk certified 11, but... for the life of me I can't find the latest release on downloads.asterisk.org |
| 12:58.01 | protocoldoug | the regular asterisk folder, everything is updated as of yesterday, but... the certified folder, nothing updated since march |
| 12:59.09 | WIMPy | wmchris: Could it be that it's just tehe callerID name, not the number? |
| 13:00.06 | *** join/#asterisk dgeary2 (~debian@pa49-187-94-200.pa.nsw.optusnet.com.au) |
| 13:07.48 | *** join/#asterisk vlad_starkov (~vlad_star@194.154.71.230) |
| 13:08.24 | *** join/#asterisk CIDname (~CIDname@c-76-30-153-192.hsd1.tx.comcast.net) |
| 13:09.43 | PLMg | hello, anyone willing to give a hand on configuring sendmail? |
| 13:10.02 | bacobart | how come when you use originatecall via ami not both the src and dst are filled in in the cdr database? |
| 13:11.07 | Greenlight | bacobart: Are you specifying a callerid in the AMI Originate ? |
| 13:11.46 | bacobart | yes i am |
| 13:12.29 | Greenlight | Then I'm not sure, sorry. I'm pretty certain they've always been populated for me |
| 13:12.43 | *** join/#asterisk davlefouAMD (~david@197.15.217.90) |
| 13:13.18 | bacobart | hm |
| 13:13.36 | Greenlight | Which version are you using ? |
| 13:13.38 | bacobart | maybe you're right, the callerid isn't always populate i'm noticing now |
| 13:14.05 | bacobart | i'm using v11, but i think you're on to something. probably my mistake. thanks. |
| 13:14.11 | Greenlight | Okay :) |
| 13:14.23 | *** join/#asterisk serafie (~erin@nat/digium/x-hvlqlbwegnfauluo) |
| 13:16.34 | *** join/#asterisk newtonr (~newtonr@nat/digium/x-ztexfgtvrheibqjp) |
| 13:16.34 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 13:22.51 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 13:26.50 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-knhbynqveqbevfta) |
| 13:26.50 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:32.21 | boom^time | Hey guys, what's a good way to force all of my SIP traffic to replace my LAN IP with my WAN IP? |
| 13:32.39 | *** join/#asterisk russum (~russum@ool-4a599ffe.dyn.optonline.net) |
| 13:32.41 | *** part/#asterisk russum (~russum@ool-4a599ffe.dyn.optonline.net) |
| 13:33.24 | bacobart | boom^time; externip |
| 13:33.34 | boom^time | bacobart, thanks |
| 13:35.29 | boom^time | hmm I put that in sip.conf [general] and under the trunk but it's still sending my nat IP |
| 13:36.20 | Greenlight | Do you have nat=yes and localnet set ? |
| 13:36.56 | boom^time | I have nat=force_rport,comedia |
| 13:37.07 | boom^time | and I'm not aware of the localnet option |
| 13:37.24 | [TK]D-Fender | what is a NAT IP? |
| 13:37.37 | [TK]D-Fender | So far we heard of LAN IP and WAN IP... what's this 3rd IP? |
| 13:37.54 | boom^time | sorry I meant LAN ip |
| 13:37.59 | [TK]D-Fender | [09:36]boom^timeI have nat=force_rport,comedia <- this is only for reinvites, not for contact IP's |
| 13:38.01 | boom^time | NAT IP=LAN IP |
| 13:38.57 | [TK]D-Fender | First, why would you need to set all comms to use WAN only? |
| 13:40.26 | boom^time | because I have the necessary ports forwarded to my dev machine for asterisk to work and the trunk I'm using is expecting my public IP |
| 13:40.41 | Greenlight | YOu shouldn't need to forward any ports |
| 13:41.08 | [TK]D-Fender | yes, he should |
| 13:41.15 | boom^time | how is 5060 going to get in? along with all of the RTP ports? |
| 13:41.21 | [TK]D-Fender | * behind NAT should have ports forwarded |
| 13:41.28 | Greenlight | NAT punches a hole |
| 13:41.31 | [TK]D-Fender | boom^time: and you don't force WAN on all comms, just EXTERIOR comms |
| 13:41.57 | boom^time | Right, otherwise it'l speak that way to my local devices. |
| 13:42.08 | Greenlight | Oh, he's got users external, for some reason I presumed it was just his ITSP that was external |
| 13:42.09 | Greenlight | My bad. |
| 13:42.12 | [TK]D-Fender | boom^time: So set your localnet ranges, externip, and set each peer's NAT settings based on their circumstances (which is usually "no") |
| 13:43.08 | igcewieling | boom^time: it has been at least a week and you STILL have not read sip.conf.sample? |
| 13:43.53 | igcewieling | it has a whole fscking section on NAT and localnet and externaddr, etc. |
| 13:45.02 | [TK]D-Fender | I should write that mini-book I've bee putting off.... |
| 13:46.16 | igcewieling | [TK]D-Fender: I already wrote one. Here it is: Asterisk Book, "core show functions", "core show applications", read all relevant .sample files. |
| 13:46.58 | [TK]D-Fender | igcewieling: No... I mean a real guide with real clear information. |
| 13:47.12 | igcewieling | [TK]D-Fender: nobody will read that either. 8-| |
| 13:47.14 | [TK]D-Fender | igcewieling: not "sift through files and hope you see the big picture" |
| 13:47.23 | [TK]D-Fender | igcewieling: They will if it's written propery |
| 13:47.31 | [TK]D-Fender | properly* |
| 13:47.49 | igcewieling | we can't even get people to read what you write here on the channel. 8-( |
| 13:48.01 | boom^time | [TK]D-Fender, that worked, I'm still getting congestion failures but I believe that is my trunks issue. So waiting for their response on my trace. |
| 13:48.27 | Greenlight | boom^time: You're specifying an allowed codec ? |
| 13:48.32 | [TK]D-Fender | boom^time: How about showing us? |
| 13:48.47 | [TK]D-Fender | Greenlight: Codecs have nothing to do with congestion |
| 13:49.15 | [TK]D-Fender | Not that I believe that is actually the pertinent response |
| 13:49.34 | Greenlight | I wasn't sure what he meant when he said congestion, so thought i'd check.. |
| 13:49.48 | boom^time | http://pastebin.com/pyAPdDQG |
| 13:49.52 | [TK]D-Fender | Greenlight: Key is that nothing suggests codecs |
| 13:50.29 | Greenlight | Oh, 401 Unauthorized |
| 13:51.05 | [TK]D-Fender | boom^time: Fix your auth on your peer |
| 13:51.32 | [TK]D-Fender | boom^time: From: "501" <sip:501@75.151.20.174>;tag=as327a0162 <- you are also passing on the callerID from your originating channel.. they probably don't ID you because of this as well |
| 13:52.09 | [TK]D-Fender | boom^time: "fromuser=username" <- Also.... your provider is NOT behind NAT.... they should be "nat=no" as well |
| 13:52.40 | [TK]D-Fender | boom^time: The only typical time you set "nat=yes" for a peer entry is for individual phones behind another NAT |
| 13:53.32 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:53.32 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:53.33 | boom^time | [TK]D-Fender, they don't auth based on CID. My auth info is exactly as they provided. |
| 13:53.54 | [TK]D-Fender | boom^time: make the change |
| 13:53.55 | igcewieling | boom^time: then ask them why they are rejecting your call. |
| 13:53.58 | boom^time | I'm going to turn the NAT off, I think I had it reversed in my mind, where since I'm natted I should specify that nat is necessary |
| 13:54.16 | [TK]D-Fender | boom^time: Why would you specify YOUR situation on a peer by peer basis? |
| 13:54.17 | igcewieling | boom^time: nat= only refers the endpoint |
| 13:54.39 | [TK]D-Fender | boom^time: it's a PEER setting in there.. YOUR'S is in [general] |
| 13:55.27 | boom^time | Oh wow yeah that makes so much more sense. |
| 13:56.37 | boom^time | Still congested. Time to bug them. |
| 13:56.41 | [TK]D-Fender | show us |
| 13:57.05 | [TK]D-Fender | and never say "congested". that is not the error. |
| 13:57.18 | [TK]D-Fender | it previosy got refused on a 401 |
| 13:57.31 | boom^time | still unauthorized* |
| 13:57.36 | [TK]D-Fender | ... |
| 13:57.38 | [TK]D-Fender | PASTEBIN it |
| 13:57.55 | boom^time | http://pastebin.com/yqE4tz7q |
| 13:59.36 | [TK]D-Fender | you have not done the change I told you to do |
| 14:00.06 | boom^time | But I did. I added nat=no under the peer sections |
| 14:00.15 | [TK]D-Fender | A[09:52][TK]D-Fenderboom^time: "fromuser=username" <- Also.... your provider is NOT behind NAT.... they should be "nat=no" as well |
| 14:01.01 | *** part/#asterisk PLMg (PLMg@78.96.151.225) |
| 14:01.38 | *** join/#asterisk MLNoah (~chatzilla@noc.metalink.net) |
| 14:03.11 | boom^time | with fromuser=username under the peer |
| 14:03.11 | boom^time | http://pastebin.com/fY4EYS4i |
| 14:04.39 | [TK]D-Fender | pastebin the peer, mask only the SECRET |
| 14:04.58 | [TK]D-Fender | also, have any calls been successful through this peer in the past? |
| 14:05.15 | boom^time | http://pastebin.com/taFUrg5h |
| 14:05.20 | boom^time | No |
| 14:05.46 | [TK]D-Fender | externip=75.151.20.174 <- this belongs under [general] only |
| 14:05.52 | boom^time | got it |
| 14:05.52 | [TK]D-Fender | this is not a per-peer setting |
| 14:06.08 | igcewieling | also it isn't fromuser=username |
| 14:06.37 | [TK]D-Fender | user=vpsdresult2 c<- this SHOULD be username, not "user" |
| 14:06.40 | igcewieling | fromuser=theusernameyouritspexepcts. |
| 14:06.52 | [TK]D-Fender | multiple mistakes there |
| 14:06.53 | boom^time | oh jesus. |
| 14:07.04 | boom^time | I'm not with it this morning |
| 14:07.31 | boom^time | fixed them both |
| 14:07.53 | igcewieling | checks boom^time's karma, oh, dropped 3 points in the last hour. |
| 14:08.10 | boom^time | Yeah I understand. |
| 14:08.14 | [TK]D-Fender | boom^time: also "type=peer" |
| 14:09.01 | boom^time | [TK]D-Fender, doesn't that limit it to outbound only? or do I have the peer/friend/user understanding wrong |
| 14:09.13 | boom^time | There are DIDs that will be coming in from that as well |
| 14:09.22 | [TK]D-Fender | boom^time: Your understanding is wrong... |
| 14:09.53 | igcewieling | boom^time: your understanding is based on very old and outdated information which has not been accurate since Asterisk 1.2 in 2002 |
| 14:09.54 | [TK]D-Fender | make all changes. retest then PB it all again on failure. |
| 14:10.13 | [TK]D-Fender | igcewieling: Date is a little off :) |
| 14:10.24 | igcewieling | I believe it may be covered in the UPGRADE* files which you didn't read. |
| 14:10.30 | [TK]D-Fender | We weren't at whole numbers until about 2004 :) |
| 14:10.50 | *** join/#asterisk Blashyrkh (~chatzilla@043-054-094-081.as39912.net) |
| 14:11.04 | Blashyrkh | i have a question regarding dialplan syntax |
| 14:11.09 | igcewieling | [TK]D-Fender: You might be right. 1.4 came out around Katrina in 2005 |
| 14:11.46 | [TK]D-Fender | Blashyrkh: Ask away |
| 14:11.54 | boom^time | http://pastebin.com/9SAFAG5z |
| 14:12.02 | Blashyrkh | still trying to figure out what exactly :D |
| 14:12.33 | Blashyrkh | exten => _X.,n+1(trunk_routing),Gotoif($[${EXTEN:0:8} = "certainnumber "]?mgw4) |
| 14:12.45 | [TK]D-Fender | Blashyrkh: tehre is no "n+1" |
| 14:12.57 | [TK]D-Fender | Blashyrkh: That is not legal |
| 14:12.58 | Blashyrkh | the thing behind the questino mark in the goto is the thing that gets executed when the clause is true? |
| 14:13.27 | Blashyrkh | are you sure, i am jsut editing an existing dialplan |
| 14:13.37 | boom^time | BlackDex, it's a label for a priority, it jumps to it |
| 14:13.40 | igcewieling | BlackDex: Hint: when there are quotes on one side of an = you almost ALWAYS need quotes on the other side of the = |
| 14:13.40 | boom^time | argh |
| 14:13.53 | boom^time | Blashyrkh* |
| 14:14.19 | BlackDex | darn highlights :p |
| 14:14.37 | [TK]D-Fender | boom^time: all <- |
| 14:15.07 | Blashyrkh | hm? |
| 14:15.16 | Greenlight | When did the n+1 syntax get depreacted? |
| 14:15.29 | igcewieling | Blashyrkh: Hint: when there are quotes on one side of an = you almost ALWAYS need quotes on the other side of the = |
| 14:15.29 | Blashyrkh | its a really old installation |
| 14:15.33 | boom^time | [TK]D-Fender, did I leave something out? |
| 14:15.37 | Blashyrkh | the quotes are jsut a number |
| 14:15.42 | [TK]D-Fender | boom^time: the new CONFIGS |
| 14:15.46 | Blashyrkh | didnt know how to put it, and didnt want to post my number :D |
| 14:16.13 | igcewieling | Blashyrkh: the line you pasted will NEVER be true. |
| 14:16.30 | boom^time | [TK]D-Fender, http://pastebin.com/xPcvfx5d |
| 14:16.38 | igcewieling | ${EXTEN:0:8} will NEVER be equal to "certainnumber " |
| 14:16.42 | Blashyrkh | Gotoif($[${EXTEN:0:8} = 1234556]?mgw4) |
| 14:16.48 | Blashyrkh | thats what i shouldve put here :D |
| 14:17.01 | igcewieling | Blashyrkh: or just put quotes on both sides of the = |
| 14:17.04 | igcewieling | pretty simple. |
| 14:17.04 | Blashyrkh | i jsut deleted my number and the "certainnumber" is a aplaceholder for a random number |
| 14:17.38 | Blashyrkh | but if the expression is true it will do the thing behind the questino mark right? |
| 14:17.41 | igcewieling | Blashyrkh: NEVER EVER mask or change anything when pasting to the channel EXCEPT for passwords. You will waste everyone's time. |
| 14:17.49 | Blashyrkh | okay |
| 14:17.57 | igcewieling | Blashyrkh: yes and you'll see a 1 in the CLI in place of the expression |
| 14:18.25 | [TK]D-Fender | boom^time: user=vpsdresult2 <- I told you this parameter name does not exist. It should not be there |
| 14:18.39 | [TK]D-Fender | boom^time: So who is this service you are connecting to?\ |
| 14:20.01 | boom^time | [TK]D-Fender, just commented it out, reload, still error. They are michigan network services |
| 14:20.42 | [TK]D-Fender | boom^time: The rest of the peer looks sane but I don't know if they require fromdmain being set, or you having registered first, etc. |
| 14:20.51 | [TK]D-Fender | boom^time: Go check their instructions |
| 14:21.46 | boom^time | [TK]D-Fender, will do and thanks for all of your help. Although I'm apparently incapable of following instructions this morning. I'm moving this week and have had 0 sleep. Need to go make some more coffee. |
| 14:24.42 | Blashyrkh | iam unsure about the scope of priorites |
| 14:25.15 | Blashyrkh | if the priority is labeled with the same name multiple times in multiple includes |
| 14:25.27 | Blashyrkh | how can i see which one a certain statement will jump to |
| 14:27.43 | [TK]D-Fender | Blashyrkh: You are not supposed to have duplicat labels on an exten |
| 14:33.10 | boom^time | [TK]D-Fender, the ITSP came back and corrected my peer, it's now working: http://pastebin.com/DiUiX4Uk |
| 14:34.32 | [TK]D-Fender | boom^time: user=vpsdresult2 <- 3rd time now. GET RID OF THIS |
| 14:35.23 | [TK]D-Fender | boom^time: seems defaultuser has supplanted username completely. |
| 14:35.24 | boom^time | [TK]D-Fender, I removed it |
| 14:35.33 | boom^time | [TK]D-Fender, they came back and readded it |
| 14:39.11 | kaldemar | ll |
| 14:39.41 | *** join/#asterisk vNistelroot (~secure@68.Red-83-60-117.dynamicIP.rima-tde.net) |
| 14:39.58 | boom^time | [TK]D-Fender, yeah it's not until I add defaultuser that it works. |
| 14:40.53 | vNistelroot | hi ! ive received notification about the last vulns, ive to deploy the new full version? or could I patch the affected deployed pbx ? |
| 14:41.14 | boom^time | I removed the user entry it definitely does not need to be there. |
| 14:42.17 | *** join/#asterisk Rumbles (~Rumbles@host81-133-199-16.in-addr.btopenworld.com) |
| 14:49.19 | igcewieling | vNistelroot: wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.5.1.tar.gz && tar -zxvf asterisk-11.5.1.tar.gz && cd asterisk-11.5.1 && ./configure && make install && make config |
| 14:50.39 | *** join/#asterisk vlad_starkov (~vlad_star@109.95.84.116) |
| 14:52.41 | *** join/#asterisk nix8n82 (~AndChat27@24.143.11.81) |
| 14:54.24 | *** part/#asterisk ghost75 (~trechber@dslb-088-064-052-146.pools.arcor-ip.net) |
| 14:56.37 | *** join/#asterisk ghost75_ (~quassel@dslb-088-064-052-146.pools.arcor-ip.net) |
| 14:57.44 | Katty | peeks in |
| 14:58.23 | [TK]D-Fender | pokes out |
| 14:58.30 | *** join/#asterisk vlad_starkov (~vlad_star@109.95.84.116) |
| 14:59.21 | Katty | [TK]D-Fender: how're you dear? did you get your xray results back yet? |
| 15:00.43 | vNistelroot | Hi igcewieling , thanks for the info !, i only want to be on the safe side ( no break config and so on ) |
| 15:01.00 | vNistelroot | should I take care of anything ? |
| 15:05.38 | *** join/#asterisk _Corey_ (~chatzilla@50-200-184-54-static.hfc.comcastbusiness.net) |
| 15:07.10 | [TK]D-Fender | Katty: Doc didn't even take one. hardly did anything. just looked at the overall chest alignment, heard my report that I had a significant bone movement from the previous fusion point the night before and said I'm in a sling for another 3 weeks, physio in 4, and back for checkup in 8 |
| 15:07.22 | *** join/#asterisk vlad_starkov (~vlad_star@109.95.84.116) |
| 15:07.25 | [TK]D-Fender | Katty: basically very little to say |
| 15:08.00 | jmetro | [TK]D-Fender: doctors try to do as little work as possible |
| 15:10.12 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 15:11.12 | *** join/#asterisk vlad_starkov (~vlad_star@109.95.84.116) |
| 15:14.21 | Katty | [TK]D-Fender: hrmmm |
| 15:14.36 | Katty | [TK]D-Fender: well, i hope everything turns out ok |
| 15:16.21 | *** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa) |
| 15:18.11 | *** join/#asterisk felipealmeida (~user@177.17.117.52) |
| 15:23.40 | *** join/#asterisk navaismo (~navaismo@189.241.19.115) |
| 15:27.24 | Vann | Is there a way to determine whether asterisk is running in 64 or 32bit mode? Or is it entirely tied to the OS? |
| 15:30.39 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:35.34 | Kobaz | there is no 'mode' |
| 15:35.39 | Kobaz | file /usr/sbin/asterisk |
| 15:35.40 | newtonr | Vann, The OS. There is no mode |
| 15:35.50 | Kobaz | it'll tell you if it's 32 or 64 |
| 15:36.10 | Vann | gotcha, thanks |
| 15:36.37 | Vann | for some reason I was under the impression there were 32bit and 64bit binaries |
| 15:36.44 | Kobaz | there can be |
| 15:36.50 | Kobaz | it depends what you put on your system |
| 15:36.54 | Vann | ahh |
| 15:36.59 | Vann | hmm |
| 15:37.15 | Kobaz | you should get something like this: /usr/sbin/asterisk: ELF 32-bit LSB executable, Intel 80386, version 1 (SYSV), dynamically linked (uses shared libs), for GNU/Linux 2.6.18, not stripped |
| 15:37.18 | Vann | there did not appear to be a distinction in the source files |
| 15:37.29 | Kobaz | Vann: there wont be any distinction in the source |
| 15:37.34 | Vann | I gotcha |
| 15:37.40 | Kobaz | the source is the source |
| 15:37.48 | Kobaz | you can build for 8 bit systems if you really wanted to |
| 15:38.11 | Vann | Makes sense |
| 15:38.14 | Kobaz | and this is what 64 looks like: /usr/sbin/asterisk: ELF 64-bit LSB executable, x86-64, version 1 (SYSV), dynamically linked (uses shared libs), for GNU/Linux 2.6.26, not stripped |
| 15:38.37 | Vann | Aha, thank you. |
| 15:38.44 | Kobaz | a quick 'file /usr/sbin/asterisk' will tell you in seconds |
| 15:44.48 | *** join/#asterisk ChadAragorn (~ChadArago@63-235-131-194.dia.static.qwest.net) |
| 15:46.17 | *** join/#asterisk Snipper_ (~IceChat9@mail.aytycrm.com.br) |
| 15:48.19 | Katty | grooves |
| 15:52.47 | *** join/#asterisk davlefouAMD (~david@197.15.217.90) |
| 15:59.43 | SeanN | WIMPy, another question re our previous discussion.... |
| 16:00.03 | SeanN | if I want to send a call to a Misc Destination, how would I do that? |
| 16:00.28 | SeanN | what would the syntax be assuming the feature code is 3838? |
| 16:00.51 | [TK]D-Fender | SeanN: #freepbx <------- |
| 16:01.02 | SeanN | "Exten: SIP/3838" doesn't work, I guess since it's not a SIP device |
| 16:01.31 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.3) |
| 16:01.54 | [TK]D-Fender | SeanN: Local/3838@from-internal/n |
| 16:02.06 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 16:02.13 | WIMPy | If you want to go via dialplan you need to use a local channel. |
| 16:22.16 | *** join/#asterisk SirLagz (~sirlagz@ppp121-45-232-106.lns20.per1.internode.on.net) |
| 16:23.47 | SeanN | this helped me a lot (again), thanks! |
| 16:26.35 | WIMPy | Is there anything special about using ! patterns with realtime? Do they need to be escaped? |
| 16:26.55 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:91bf:8d0c:a509:62a8) |
| 16:28.34 | *** join/#asterisk dongola7 (~dongola7@50-197-3-153-static.hfc.comcastbusiness.net) |
| 16:38.44 | Katty | so quiet. |
| 16:39.42 | WIMPy | Katty: Talk to j4jackj. That should make for some traffic. |
| 16:40.09 | Katty | hmm. |
| 16:40.14 | Katty | possibly. if the krave isn't gone. |
| 16:41.11 | ChannelZ | yawns |
| 16:44.53 | *** join/#asterisk btcquant (~noahnoah@99-41-172-179.lightspeed.irvnca.sbcglobal.net) |
| 16:45.49 | Katty | ChannelZ: o/ |
| 16:47.58 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 16:48.17 | jmetro | Katty: we require more vespene gas. |
| 16:48.51 | Katty | puts on the gloves. |
| 16:49.36 | *** part/#asterisk SeanN (~Sean_Nien@cpc1-hers1-0-0-cust582.6-3.cable.virginmedia.com) |
| 16:49.59 | Katty | we're goin to Antiga Prime! |
| 16:50.06 | Katty | fires up the engine |
| 16:51.44 | jmetro | Antiga Prime? Stop by the citadel :3 |
| 16:55.33 | jmetro | mass effect engines are the only way terrans ever reached protoss space. |
| 16:55.48 | *** join/#asterisk NoSu (~TeknoJuce@xbmc/staff/TeknoJuce) |
| 16:56.52 | NoSu | hi I have context called outbound that is able to call through google voice, but now I would like it to also be able to dial internal extensions how would I go about doing this |
| 16:58.04 | [TK]D-Fender | NoSu: Put extens in there to dial the other stuff you want |
| 16:58.14 | protocoldoug | In the security release from last night, there was a mention that there was an asterisk-11.2-cert2, but the link in the email is broken to: http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.2-cert2 -- am I looking in the wrong place? |
| 16:59.09 | [TK]D-Fender | there is no cert2 there |
| 17:00.15 | Qwell | mjordan: ^^? |
| 17:00.17 | protocoldoug | exactly, yeah i was hoping to find an update to certified-asterisk 11, last update being in march, none from last night. |
| 17:00.25 | *** join/#asterisk PLMg (~L@78.96.151.225) |
| 17:00.50 | NoSu | [TK]D-Fender, in extensions what would something like that look like? |
| 17:00.51 | mjordan | hum hum hum |
| 17:01.11 | [TK]D-Fender | mjordan: Can you fix your dates in the topic as well.... |
| 17:01.24 | mjordan | lawl |
| 17:01.36 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.5.1 (2013/08/27), 10.12.3 (2013/08/27), 1.8.23.1 (2013/08/27), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 17:01.50 | mjordan | that was a long day. There is an 11.2-cert2; I'll find out what happened to it |
| 17:01.51 | [TK]D-Fender | NoSu: exten => somthing,1,Dial(something) |
| 17:01.53 | [TK]D-Fender | NoSu: ... |
| 17:01.55 | [TK]D-Fender | ~book |
| 17:01.55 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 17:01.57 | [TK]D-Fender | ^^^ |
| 17:02.02 | protocoldoug | thank you mjordan!! |
| 17:02.10 | [TK]D-Fender | NoSu: "core show application dial" |
| 17:02.49 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 17:03.59 | NoSu | if I have 10 phones I would have to put all 10 extensions in there or is there like a wildcard of sorts |
| 17:04.33 | [TK]D-Fender | NoSu: Or something like that |
| 17:04.45 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 17:05.23 | mjordan | heh. Forgot to press the big red release button on that one. protocoldoug: should be just a couple of seconds. |
| 17:05.50 | mjordan | protocoldoug: check now |
| 17:06.10 | protocoldoug | mjordan: o/ dude, you rock. thank you a ton :) |
| 17:06.13 | NoSu | So I would have to basically put 100 (10x10) combinations of ext dialing in the extensions.conf |
| 17:06.26 | mjordan | np, thanks for finding that and letting us know! |
| 17:06.30 | Greenlight | NoSu: You can pattern match |
| 17:06.38 | protocoldoug | no prob, too! |
| 17:06.53 | Greenlight | exten => _XXX,1,NoOp(Hello) |
| 17:06.57 | paulc | NoSu: exten => _2XX,1,Dial(SIP/${EXTEN}) |
| 17:06.59 | Greenlight | WOuld match all 3 digit extensiopns |
| 17:07.05 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 17:07.57 | paulc | although I generally prefer defining each extension explicitly |
| 17:09.42 | NoSu | so you would write 1000 enteries if you had 1000 phones |
| 17:10.04 | Greenlight | I don't |
| 17:10.11 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 17:10.33 | paulc | I might. Depending on how it was set up. The last system we did we auto generate sip.conf and extensions.conf fragments from a database, because different users have different feature sets. |
| 17:10.46 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 17:10.48 | paulc | If everyone's mostly the same, you could pattern match the bulk of them, and write the exceptions individually |
| 17:10.57 | Greenlight | Yea, that's the way FreePBX does it as well. BUt it's not the most efficient. |
| 17:11.23 | NoSu | have you guys ever used a video door phone? |
| 17:11.47 | Greenlight | My old flat had one... |
| 17:11.54 | NoSu | how are they handled in asterisk? |
| 17:11.54 | Greenlight | Does that count ? :) |
| 17:12.04 | Greenlight | Oh - it wasn't connected to asterisk. |
| 17:12.25 | Greenlight | BUt i guess it would depend on the phone |
| 17:13.04 | NoSu | basically you press the button on it and I would assume that that would be like a speed dial button or something to ring the phones |
| 17:13.13 | NoSu | does that sound right? |
| 17:14.54 | [TK]D-Fender | NoSu: Something like that |
| 17:15.04 | [TK]D-Fender | NoSu: It dials what you tell it to dial |
| 17:15.47 | NoSu | is there any uber cheap model that you know of that is nice to asterisk? |
| 17:16.15 | Blashyrkh | what is a good place to talk about sip |
| 17:17.35 | igcewieling | Blashyrkh: Have you tried #gentoo? |
| 17:17.51 | NoSu | I saw dell make a horrible conflicting acronym the other day they call their KVM cables SIP cables (USB Server Interface Pod Cables) |
| 17:18.28 | NoSu | someone kept trying to explain to me what a "SIP Cable" was and I thought they were off their rocker |
| 17:18.37 | NoSu | http://accessories.us.dell.com/sna/productdetail.aspx?c=us&l=en&s=bsd&cs=04&sku=430-4346 |
| 17:18.55 | [TK]D-Fender | The IT world is full of acronym overlap |
| 17:19.05 | igcewieling | NoSu: you won't make much progress until you read The Book |
| 17:19.15 | Blashyrkh | iam talking about the SIP Protocol |
| 17:19.26 | Katty | i'll talk about your sip protocol in a minute. |
| 17:19.29 | igcewieling | Blashyrkh: and yet this channel is for Asterisk. |
| 17:19.37 | Katty | and recipes. |
| 17:19.48 | _Corey_ | Blashyrkh: Everyone likes to talk about the SIP protocol... there is no bad place |
| 17:19.49 | Katty | and trolling. |
| 17:19.52 | Blashyrkh | :D |
| 17:20.03 | Katty | we also talk about _Corey_ when he's not here. |
| 17:20.27 | _Corey_ | :) |
| 17:20.40 | igcewieling | For the most part there are only a few people crazy enough to have technical discussions about a protocol the Marquis de Sade would be proud of. |
| 17:20.54 | Katty | and i'm not one of them. |
| 17:21.07 | Blashyrkh | haha |
| 17:21.08 | igcewieling | Katty: most people here are not one of those people. |
| 17:22.09 | Katty | pats Blashyrkh |
| 17:22.29 | Katty | Blashyrkh: we can talk about sip anyway, if you want. |
| 17:22.33 | Katty | Blashyrkh: isn't it fancy? |
| 17:22.42 | Blashyrkh | i have a question about early media |
| 17:22.53 | Katty | well back in the day... |
| 17:22.56 | Katty | there wasn't nearly as much of it |
| 17:23.02 | Katty | media, i mean. |
| 17:23.04 | Blashyrkh | iam not quite sure who initiates it |
| 17:23.16 | Katty | well traditionally the boy would. |
| 17:23.17 | Greenlight | The person you're calling, via a 183 |
| 17:23.27 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:23.27 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:23.38 | Blashyrkh | but in what cases |
| 17:24.01 | Greenlight | There's no "set rules" |
| 17:24.01 | Blashyrkh | i have two traces here to the same endpoint, in one case it initates a 183 early media |
| 17:24.02 | _Corey_ | in whatever case they're configured to do so |
| 17:24.10 | Blashyrkh | but in the other not |
| 17:24.11 | Greenlight | Indeed |
| 17:24.13 | Greenlight | That's okay |
| 17:24.22 | Greenlight | SOme send 183 media and 180 ringing |
| 17:24.26 | Greenlight | Some send only 180 ringing |
| 17:24.32 | Greenlight | Both are valid |
| 17:24.56 | Greenlight | I find when I'm calling to a PSTN endpoint I get 183 early media |
| 17:25.19 | _Corey_ | Blashyrkh: there's obviously something about your invite causing the far end to respond with a 183 rather than a 180 |
| 17:25.21 | Greenlight | And if I'm calling a SIP endpoint I get genarlly just 180 rining |
| 17:25.43 | Blashyrkh | corey exactly, and i am thinking about what could that be |
| 17:25.48 | Greenlight | The same desination ? |
| 17:25.55 | Blashyrkh | yes |
| 17:26.04 | Greenlight | And what *is* that early media ? |
| 17:26.22 | _Corey_ | typically music on hold or something |
| 17:26.31 | Greenlight | FOr me it's usually just ringing |
| 17:26.32 | _Corey_ | "session progress" sounds... |
| 17:26.55 | NoSu | they should put more games of zork on hold instead of music |
| 17:27.01 | _Corey_ | whoops, Greenlight I thought you were Blashyrkh there for a sec |
| 17:27.08 | _Corey_ | I'm not paying attention |
| 17:27.10 | Greenlight | :) |
| 17:27.37 | Blashyrkh | i just have following problem, iam calling the same destination, which is answering with a voice recording |
| 17:27.53 | Greenlight | Eraly media is BEFORE the answer... |
| 17:27.55 | Blashyrkh | in one case, where i see that no early media is initated, i cant here it |
| 17:28.10 | Blashyrkh | i think its planned to be before the answer, so as not to cost |
| 17:28.21 | Greenlight | Yea, that's unreliable for exactly this reaason. |
| 17:28.39 | igcewieling | Blashyrkh: if is they are idiots since early media is supposed to be ONE way. |
| 17:28.56 | Blashyrkh | its okay its supposed to be from the endpoint iam calling to me? |
| 17:29.00 | _Corey_ | it's more common than you think |
| 17:29.18 | Blashyrkh | but now iam thinking about why iam not getting early media in the other case |
| 17:29.22 | Greenlight | Yes, one way FROM the endpoint TO you. |
| 17:29.26 | igcewieling | _Corey_: I know. I never have found a good solution. |
| 17:29.46 | Greenlight | Blashyrkh: Its upto your endpoint really... |
| 17:30.23 | igcewieling | Blashyrkh: you understand that calling the same number very may not be handled by the same equipment |
| 17:30.29 | Greenlight | +1 |
| 17:30.47 | Greenlight | Might be your ITSP load balacning accross differently configured switches for all you know, |
| 17:30.49 | igcewieling | When you call us, your call could hit one of three servers. when calling out our calls can hit any of a dozen servers at the carrier. |
| 17:30.56 | _Corey_ | igcewieling: I remember spending like 4 hrs about 5 years ago figuring out why I couldn't call US Airways 800# from my desk... I had a Dial() 'r' argument and they never answered until you push a button. *doh* |
| 17:31.03 | Blashyrkh | i think it is in this case, and the effect is very reliable |
| 17:31.32 | Greenlight | How is it reliable ? |
| 17:31.56 | igcewieling | _Corey_: "r" on Dial? You must have been a TOTAL n00b at the time. |
| 17:32.00 | Blashyrkh | everytime i call with one pbx i get early media and the voice, and vice versa |
| 17:32.13 | Blashyrkh | its not like its random because of load balancing or round robin |
| 17:32.21 | Greenlight | So this is one TWO PBX's ? |
| 17:32.25 | Greenlight | NOW you say this. |
| 17:32.29 | Blashyrkh | ? |
| 17:32.32 | _Corey_ | igcewieling: We all start somewhere... ;) |
| 17:32.35 | igcewieling | Blashyrkh: remember the PBX is likely NOT running SIP. |
| 17:32.57 | Blashyrkh | no the call i make is going out fom two different |
| 17:33.00 | Blashyrkh | asterisk installations |
| 17:33.12 | Blashyrkh | iam calling the same number |
| 17:33.17 | igcewieling | then say Asterisk. when you say PBX I assume some random PBX at some random company |
| 17:33.19 | Blashyrkh | but am not getting the same effect |
| 17:33.31 | igcewieling | Blashyrkh: then you have some configuration difrerence |
| 17:33.51 | Blashyrkh | obviously :D, thats what iam looking for |
| 17:34.38 | Blashyrkh | are there no "rules of thumb, when early media is answered? |
| 17:36.56 | Blashyrkh | should there be a 180 or 183 between the 100trying and the 200ok? |
| 17:37.07 | *** part/#asterisk btcquant (~noahnoah@99-41-172-179.lightspeed.irvnca.sbcglobal.net) |
| 17:37.19 | *** join/#asterisk btcquant (~noahnoah@99-41-172-179.lightspeed.irvnca.sbcglobal.net) |
| 17:37.23 | *** part/#asterisk btcquant (~noahnoah@99-41-172-179.lightspeed.irvnca.sbcglobal.net) |
| 17:40.50 | igcewieling | Blashyrkh: "scp /etc/asterisk/sip.conf root@theotherserverip:/etc/asterisk" then reload Asterisk on theotherserverip now you have identically configured Asterisks. all you have to do is verify they are the EXACT SAME VERSION of Asterisk on the two servers. |
| 17:41.13 | igcewieling | might want to back up the sip.conf on theotherserver first though. |
| 17:45.04 | NoSu | Which android app would people recommend for asterisk use, CSipSimple sound ok? |
| 17:45.25 | ChannelZ | Yeah it's pretty decent for me |
| 17:45.48 | NoSu | ok thank you |
| 17:46.32 | ChannelZ | there are quirks here and there depending on your device |
| 17:48.28 | j4jackj | =/win 24 |
| 17:48.34 | NoSu | Galaxy Note 2 |
| 17:50.59 | ChannelZ | I don't know about that one specifically. But a 'for instance' is my EVO 3D has pops and clicks in the audio unless I set the clock rate to 32khz |
| 17:51.11 | *** join/#asterisk felipealmeida (~user@177.98.5.7) |
| 17:51.23 | NoSu | hmm interesting |
| 17:51.41 | *** join/#asterisk MLNoah (~chatzilla@noc.metalink.net) |
| 17:51.56 | NoSu | how do you set the crockrate on the fly |
| 17:52.33 | cusco | hi folks |
| 17:52.48 | ChannelZ | it's just a setting in the app. Depends what codec you're using, it might be fine on its own. 8khz works too but I was playing with wideband; even though I was using 16khz codecs, setting the clock rate to 16khz gave me pops and clicks |
| 17:53.28 | NoSu | oh I thought you were talking about the cpu |
| 17:53.32 | cusco | do telco provide ... what is it called .. overlapping? - the ability to call a DID<and some digits more> and make the call come in the DID and be able to use the extra digits in the dialplan ? |
| 17:56.52 | paulc | cusco: Is that like ISDN overlap dialing? I know that's how German direct inward dialing works.. it's weird.. like 2288-0 is reception, but 2288-103 gets you extension 103 |
| 17:58.26 | paulc | (compared to other places where you'd use something like 2288-100 for reception, because you have the range 2288-100 to 2288-199, for example) |
| 17:58.57 | protocoldoug | NoSu: don't write off zoiper. it does iax2 natively, which is pretty convenient |
| 17:59.57 | [TK]D-Fender | cusco: normally telcos take a prefix part of your # and start sending that and overlap the rest until you reach a conclusion. I've only heard of this on European E1 PRI's mind you. WIMPy has much experience in this domain IIRC |
| 18:00.17 | NoSu | ok protocoldoug |
| 18:00.39 | [TK]D-Fender | protocoldoug: What product does IAX2 ... NON-Natively? |
| 18:01.01 | protocoldoug | HAHAHAH, ok.... good point. s/natively//g; :) |
| 18:02.02 | ChannelZ | ah didn't realize there was Zoiper for mobiles. |
| 18:03.40 | ChannelZ | They're a curious company.. they had a beta of Zoiper desktop with wideband codecs.. like 4 years ago.. but have never released it. |
| 18:03.55 | igcewieling | cusco: Yes. No. Depends on your phone company and country. |
| 18:04.58 | protocoldoug | mannn, that'd be a sweet zoiper feature, too. |
| 18:07.18 | igcewieling | I don't usually post random links, but Damn! http://www.reddit.com/r/PHP/comments/1l7baq/creating_a_user_from_the_web_problem/ |
| 18:07.41 | Kobaz | haha |
| 18:07.41 | Kobaz | yeah |
| 18:07.43 | Kobaz | saw that |
| 18:08.06 | *** join/#asterisk j4jackj (jack@j4jackj-1-pt.tunnel.tserv21.tor1.ipv6.he.net) |
| 18:09.11 | cusco | I'm in europe and have PRI E1 and telco's SIP |
| 18:09.17 | cusco | from several telco's |
| 18:09.39 | cusco | my question now is.. what is this feature called? to ask the telco? |
| 18:09.55 | navaismo | ChannelZ, and quote them "We don't like paid upgrades for software, we will upgrade your license for free to Zoiper 3, even if you bought a license 10 years ago." |
| 18:09.58 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 18:11.13 | j4jackj | I'm currently listening to Asterisk's default hold music... it never gets old |
| 18:11.45 | NoSu | Put in an mp3 jukebox so you can pick your song |
| 18:12.03 | j4jackj | hehe |
| 18:12.11 | ChannelZ | yeah but even their paid versions don't do wideband, that I can see.. even Speex. |
| 18:12.27 | j4jackj | Get the Ekiga softphone... |
| 18:12.30 | ChannelZ | MOH is horrible. |
| 18:12.36 | j4jackj | Why do you need Zoiper? |
| 18:12.40 | j4jackj | ChannelZ: it is not |
| 18:12.42 | Kobaz | blink is pretty nice |
| 18:13.08 | NoSu | blink? |
| 18:13.11 | ChannelZ | I don't really need any of them. Just having a conversation. |
| 18:13.16 | Kobaz | blink soft phone |
| 18:13.23 | Kobaz | open source, probably the best one |
| 18:13.24 | NoSu | its for android? |
| 18:13.30 | Kobaz | oh android umm |
| 18:13.49 | Kobaz | i use csipsimple for android.. that's probably the best one for android |
| 18:13.58 | j4jackj | he |
| 18:14.00 | j4jackj | h |
| 18:14.25 | NoSu | that was the general consensus so far or Zoiper |
| 18:15.16 | ChannelZ | does Blink let you do 'direct' SIP calls? IE not having to actually register to a proxy or asterisk and go through that |
| 18:16.02 | j4jackj | I wonder... Does anyone want to call my conference call? sip:10@99.199.11.127 and then when you hear a dial tone, 71. Or just sip:71@99.199.11.127 . You see, my DID gives you an internal dialtone when you dial in. |
| 18:16.43 | ChannelZ | have you sorted out your firewall? |
| 18:16.48 | [TK]D-Fender | I doubt your DID gives a dialtone... |
| 18:16.56 | j4jackj | It's not a true one- |
| 18:17.00 | [TK]D-Fender | your DIALPLAN processing of the call.. is another matter |
| 18:17.01 | j4jackj | I think I have |
| 18:17.04 | igcewieling | it seems that you could test your stuff by calling your number with a cell phone |
| 18:17.05 | Kobaz | ChannelZ: yeah, uncheck the 'use proxy' option |
| 18:17.59 | ChannelZ | I'll have to try it again, don't think I've run it in awhile. It seems like all the softphones want you to create an account of some sort before they let you do anything |
| 18:19.50 | ChannelZ | Ooh I had a really old one isntalled. |
| 18:20.05 | Kobaz | yeah old blink is nothing like the new one |
| 18:20.11 | ChannelZ | 0.2.7 |
| 18:20.12 | Kobaz | you may still have to add an account |
| 18:20.18 | Kobaz | but you can make it not use a proxy for all calls |
| 18:20.22 | Kobaz | yeah they are up to 0.5 |
| 18:20.32 | Kobaz | old blink was actually kinda crappy |
| 18:22.43 | ChannelZ | the UI always seemed a bit cluttery. I'll check out a version put out this year :) |
| 18:25.57 | carrar | Nice thing about blink is you can hakc the User-Agent: strring |
| 18:26.01 | carrar | hack |
| 18:26.24 | carrar | They really should make that a dialog box |
| 18:28.58 | ChannelZ | Huh. Nifty test conference at sip2sip.info, using 48khz Opus apparently |
| 18:38.22 | ChannelZ | anyone ever farted with nf_conntrack_sip for handling incoming RTP ports in through the firewall? (Not the nf_nat_sip mod which actually munges the SIP packets) |
| 18:42.41 | NoSu | what does that accomplish? |
| 18:43.00 | jmetro | ive farted but not with nf_conntrack_sip |
| 18:46.30 | ChannelZ | Well the idea is that it watches SIP traffic and will automatically setup port forwarding as needed rather than having to manually forward an arbitrary range |
| 18:47.01 | ChannelZ | Or in general allow SIP clients behind the firewall to work more easily |
| 18:47.02 | *** join/#asterisk Merlin (merlin@evendata.net) |
| 18:47.19 | Merlin | is there a standard voice prompt for "i didn't understand that response" or something like that? |
| 18:47.33 | Merlin | in asterisk-sounds that is |
| 18:47.40 | ChannelZ | wtf-are-you-talking-about.ulaw |
| 18:47.40 | igcewieling | Merlin: all listed in /var/lib/asterisk/sounds/en/*.txt |
| 18:47.44 | Merlin | k |
| 18:47.54 | [j4jackj] | [TK]D-Fender: I have finally set up dialtone processing for if you dial that number |
| 18:47.55 | Merlin | ChannelZ: i so need that |
| 18:48.06 | ChannelZ | in cheery Allison voice |
| 18:48.09 | Merlin | exactly |
| 18:48.44 | Merlin | oh sweet, it's vm-sorry |
| 18:48.56 | Merlin | however, i still would prefer wtf-are-you-talkin-bout.ulaw |
| 18:49.02 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 18:50.12 | [TK]D-Fender | go order it |
| 18:50.13 | [j4jackj] | heh |
| 18:50.23 | [j4jackj] | [TK]D-Fender: go order what? |
| 18:50.36 | Merlin | i'll start a kickstart compaign to fund the real alice recording it |
| 18:50.38 | Penguin | She'll record almost anything you want. |
| 18:50.39 | [TK]D-Fender | That recording |
| 18:50.41 | ChannelZ | Will she cuss for money? |
| 18:50.42 | [TK]D-Fender | http://www.theivrvoice.com/ |
| 18:50.54 | [j4jackj] | Very expensive.... |
| 18:51.07 | Penguin | Order it through the digium store. It isn't that expensive. |
| 18:51.08 | mjordan | I'd take that a contibution to the sounds-extra |
| 18:51.09 | ChannelZ | hmm that'd be a good domain name.. cussformoney.com |
| 18:51.12 | mjordan | contribution... |
| 18:51.31 | Penguin | I've had her record prompts for me and I don't like to spend money. |
| 18:52.03 | _Corey_ | If you're doing small recordings, the Digium store is more cost-effective |
| 18:52.08 | Penguin | Yep. |
| 18:52.16 | Penguin | $240 per hour otherwise. |
| 18:52.41 | _Corey_ | She'll do it in 1/2 hr increments though |
| 18:52.46 | [j4jackj] | eek |
| 18:53.00 | Merlin | that's worth it for the wtf-you-talkin-bout |
| 18:53.01 | [j4jackj] | Digium must be a big company |
| 18:53.03 | _Corey_ | most of our projects come in under 1/2 hr |
| 18:53.23 | Penguin | But if you just want a couple asterisk prompts, the digium store will be the best bet. |
| 18:54.04 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 18:54.50 | ChannelZ | I don't even see it |
| 18:55.03 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 18:55.53 | Penguin | They've changed the web site since the last time I ordered prompts. I'm looking for it. |
| 18:57.04 | Penguin | http://store.digium.com/voice_prompt.php |
| 18:58.02 | _Corey_ | They used to have a special website for it at one time |
| 19:00.15 | [j4jackj] | Could I care less about professional voicng of IVRs? No. |
| 19:01.07 | Penguin | Having Allison make your custom recordings allows all your prompts to match. Having several different voices sounds unprofessional most of the time. |
| 19:02.49 | [TK]D-Fender | [j4jackj]: You're too lazy to use ANY prompts so far and just shove an empty tone. |
| 19:02.59 | [TK]D-Fender | [j4jackj]: So... yeah, that's believable :) |
| 19:03.11 | ChannelZ | $12 for "what the fuck are you talking about?" |
| 19:03.27 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.14) |
| 19:03.46 | Penguin | Cheap enough, right? |
| 19:03.57 | _Corey_ | very important for your profanity to match the system, imho |
| 19:04.15 | ChannelZ | Cheaper than phone sex. I hope the pervs don't get wind of this service. |
| 19:04.41 | Kobaz | i use text to speech half the time |
| 19:04.56 | Kobaz | because it's like random custom crap specific for one customer that no one is going to care how good it sounds |
| 19:05.00 | Kobaz | it actually sounds pretty good |
| 19:05.02 | *** join/#asterisk mokmeister (~mokmeiste@86-40-246-78-dynamic.b-ras2.lmk.limerick.eircom.net) |
| 19:05.44 | newtonr | Kobaz, what engine do you use? |
| 19:06.00 | Kobaz | cepstral |
| 19:06.10 | ChannelZ | I have an ancient version of Dectalk if you want to sound like Stephen Hawking |
| 19:06.19 | *** join/#asterisk serafie (~erin@nat/digium/x-rknkgijkbkokshke) |
| 19:06.42 | Kobaz | http://www.cepstral.com/en/telephony/linux |
| 19:06.48 | MLNoah | is there any way for "core show hints" to be ordered by hint context rather than hint extension? it's a bit distracting on a multi-tenant system to have your hint contexts interspersing |
| 19:06.48 | ChannelZ | (in fact I use it for my VM at home. He's trapped in my basement.) |
| 19:06.58 | newtonr | Kobaz, yeah I remember that sounding pretty good. I saw it a bit back when I was in tech support. |
| 19:07.42 | Kobaz | oh wow |
| 19:07.43 | [TK]D-Fender | ChannelZ: Dr. Sbaitso :P |
| 19:07.44 | Kobaz | pricing went up |
| 19:07.49 | Kobaz | it used to save to files for free |
| 19:07.52 | Kobaz | now they want $200 |
| 19:07.58 | Kobaz | it is a good enough though, it's worth it |
| 19:08.06 | Kobaz | er, good enough engine |
| 19:08.31 | ChannelZ | just record the samples off the site (or do they not do that anymore either) |
| 19:09.01 | Kobaz | i have an old version that just saves to files or streams |
| 19:09.11 | Kobaz | i wonder how the new version sounds in comparison |
| 19:09.59 | ChannelZ | Is one of these voices the ones they use on that xtranormal site? |
| 19:10.16 | [j4jackj] | [TK]D-Fender: Let's just say, I actually have a prompt now, 'Please dial 8 followed by an extension number for extensions, please dial 7 followed by a conference number for conferences.' |
| 19:10.17 | Kobaz | no idea |
| 19:10.27 | ChannelZ | ...or used to apparently |
| 19:10.46 | ChannelZ | http://www.youtube.com/watch?v=DaxU0ut5tUw&feature=player_embedded#t=0 |
| 19:11.24 | ChannelZ | The woman is Allison I think. |
| 19:11.47 | Qwell | ChannelZ: definitely not |
| 19:12.00 | Qwell | ChannelZ: The Allison Cepstral voice is much better than that garbage. |
| 19:12.01 | [j4jackj] | [TK]D-Fender: I've added a prompt to direct dial in. |
| 19:12.21 | [j4jackj] | AFAIK no more empty tone |
| 19:13.10 | ChannelZ | sounds similar to me. Guy is maybe David. |
| 19:13.23 | *** join/#asterisk serafie1 (~erin@nat/digium/x-ectqlcqqrubkmcuk) |
| 19:13.36 | Qwell | ChannelZ: If that's Cepstral at all, the quality is set to something utterly shitty. |
| 19:13.38 | *** join/#asterisk Vann (~manny@c-76-97-51-58.hsd1.ga.comcast.net) |
| 19:13.42 | Qwell | Cepstral actually sounds pretty good... |
| 19:13.46 | Kobaz | yeah |
| 19:13.52 | Kobaz | it's one of the best in the cheapo price random |
| 19:14.07 | Kobaz | range |
| 19:14.22 | Kobaz | weird... my mind is thinking one thing and my fingers are typing another |
| 19:14.26 | Qwell | hell, festival sounds better than that |
| 19:14.38 | Vann | Hi guys. Is there a way to kill a SIP channel via SIP ID? Or something similar... I have this stuck channel and not sure how to "destroy" it without restarting asterisk. |
| 19:14.40 | [j4jackj] | yea |
| 19:15.31 | [j4jackj] | Vann: hangup request SIP/ [TAB] throgh the results until you get your stuck channel (It's happened to me) |
| 19:15.51 | Kobaz | oh wow |
| 19:15.56 | Kobaz | festival has gotten waaaaaay better |
| 19:16.04 | Kobaz | holy crap |
| 19:16.18 | shamil | hi, has anyone got asterisk to work with Skype (Connect)? |
| 19:16.20 | Kobaz | actually it sounds 100 times better than cepstral :P |
| 19:17.04 | ChannelZ | shamil: is that Skype's SIP product? |
| 19:17.12 | shamil | ChannelZ, yep |
| 19:18.23 | Penguin | vann: channel request hangup <your channel ID> |
| 19:18.36 | Kobaz | channel name |
| 19:18.37 | Kobaz | not id |
| 19:18.37 | [j4jackj] | Penguin: I gave the spacesaver version |
| 19:18.42 | ChannelZ | Never used it. I had SFA when they still allowed that. |
| 19:18.42 | Kobaz | ids are different |
| 19:18.52 | Qwell | Kobaz: HTS 2011 voices are pretty good |
| 19:18.55 | Vann | Yeah trying to find the corresponding channel name to the ID. |
| 19:18.59 | Kobaz | Qwell: very good |
| 19:19.22 | ChannelZ | But if they speak proper SIP no reason why it shouldn't work OK. Are you asking pre-sales or because you have it and it doesn't work? |
| 19:19.33 | Penguin | vann: You can check with sip show channels and core show channels concise to see channels that are active. |
| 19:19.38 | Qwell | Kobaz: Anna in particular is really good |
| 19:19.55 | Vann | Penguin, thanks going to try that now. |
| 19:20.58 | Kobaz | http://www.cstr.ed.ac.uk/cgi-out//ttsYf6rS9.wav |
| 19:21.08 | shamil | ChannelZ, I would assume it's not that different |
| 19:22.07 | ChannelZ | well SFA was a separate channel driver in Asterisk. |
| 19:22.21 | [j4jackj] | SFA? |
| 19:22.27 | ChannelZ | LAME, the Festival demos are censored |
| 19:22.28 | Qwell | ~skypeforasterisk |
| 19:22.28 | infobot | [skypeforasterisk] a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.digium.com/skype for details, or http://slashdot.org/story/11/05/24/2010222/Microsoft-Kills-Skype-For-Asterisk |
| 19:23.00 | *** join/#asterisk Synx|hm (~Synx@unaffiliated/synx-hm/x-1623004) |
| 19:23.52 | [j4jackj] | Hmm |
| 19:23.52 | Synx|hm | I have a SIP Trunk question, first off i think im a bit confused about the concept of trunks in an IP world... that asside, is it possible to setup an Inbound (to asterisk) SIP trunk for say 200 simul calls? |
| 19:24.05 | [j4jackj] | Synx|hm: I would recommend IAX |
| 19:24.26 | Synx|hm | Synx|hm: SIP is my only option, im actually doing this to learn more about how it works |
| 19:24.27 | ChannelZ | SIP trunk is kind of a false term |
| 19:24.33 | Penguin | SIP does not trunk, but you can create 200 different channels without too much trouble. |
| 19:24.36 | ChannelZ | But yes |
| 19:25.09 | Penguin | Think of a trunk like a pipe containing multiple calls. SIP uses a single "pipe" for each call. |
| 19:25.14 | Synx|hm | ya so thats the issue, i work in the VoIP world and a number of customers talk about SIP trunks, i think what they mean is they want to configure outbound dialing for say 200 ports without configuring 200 individual peers |
| 19:25.28 | [j4jackj] | So, pray tell, how does IAX trunking work? |
| 19:25.38 | ChannelZ | If they're all going to the same place, it's just one peer. |
| 19:25.39 | Penguin | People have been throwing around that incorrect term for ages. |
| 19:25.44 | Vann | So I tried to kill the channel and it seems to have failed. Now I see a <ZOMBIE> next to it. I don't suppose there's another way? |
| 19:25.55 | Penguin | restart asterisk |
| 19:26.02 | Vann | Yeah I figured... |
| 19:26.08 | ChannelZ | Zombies must be killed by bullet to the head. |
| 19:26.09 | Penguin | Other than that, probably not. |
| 19:26.10 | Vann | Can't really do that though. |
| 19:26.17 | Vann | lol ChannelZ |
| 19:26.25 | shamil | however, it should still be possible to set up trunking via Skype, no? |
| 19:26.36 | [j4jackj] | shamil: Yes but it will cost |
| 19:26.40 | Synx|hm | ChannelZ: ya thats what i assumed, can a single asterisk peer originate more than one call at a time? |
| 19:26.42 | Vann | Well I appreciate the help guys. Thank you. |
| 19:26.56 | shamil | [j4jackj], cost isn't a factor at the moment |
| 19:27.05 | [j4jackj] | It is |
| 19:27.12 | [j4jackj] | shamil: Skype Manager is needed |
| 19:27.14 | ChannelZ | shamil: if by trunking you mean multiple channels/calls, yes - though I'm sure you pay out the nose for it with Skype |
| 19:27.27 | ChannelZ | Synx|hm: of course |
| 19:27.55 | ChannelZ | (albeit limited by any restrictions your ITSP or whatever is on the remote end might have) |
| 19:28.22 | Penguin | IAX2 trunking works by creating a logical "pipe" between the two peers, where all the calls go through that pipe. This reduces bandwidth by not having to create overhead on every single channel as it is created and destroyed. |
| 19:28.32 | shamil | [j4jackj]. I already have Skype Manager, business acct etc |
| 19:28.46 | [j4jackj] | shamil: Now you need to buy three SIP channels... |
| 19:28.53 | [TK]D-Fender | shamil: Then set up their SIP service like they show you on their site |
| 19:29.06 | shamil | ChannelZ, well at the moment, I'm just trying to get asterisk to register |
| 19:29.12 | [j4jackj] | Three being a lucky number |
| 19:29.12 | [j4jackj] | :D |
| 19:29.30 | shamil | [j4jackj]: three? |
| 19:29.50 | [j4jackj] | shamil: if you will[ make that many calls at once yes |
| 19:29.53 | shamil | [TK]D-Fender: I need to find it, because I can't seem to find it |
| 19:30.02 | shamil | [j4jackj], I only need to make 1 simultaneous call |
| 19:30.15 | ChannelZ | Yes. Also how many licks it takes to get to the center of a Tootsie Pop if you believe the magic TV picture box. |
| 19:30.30 | [TK]D-Fender | shamil: https://support.skype.com/en/faq/FA10572/how-do-i-manage-sip-profiles |
| 19:30.31 | [j4jackj] | shamil: then buy two channerls on the off chance you'll receive a call while sending it |
| 19:30.34 | Penguin | With SIP, there is no trunk, there is no trunking. Each call has overhead. Each call is independent of the others. This is true even if you have multiple channels between your asterisk and one other peer. |
| 19:30.36 | [TK]D-Fender | shamil: Took me less that 5 seconds. |
| 19:31.29 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 19:31.37 | shamil | [TK]D-Fender, already done that |
| 19:31.47 | shamil | [j4jackj], very unlikely in this specific environment |
| 19:31.57 | shamil | not to mention I won't be answering any calls |
| 19:31.58 | Synx|hm | ChannelZ: one more question, i assume its not possible to reverse the flow here with the trunk, so say ive got 200ports on this trunk inbound to asterisk to dial out, can that trunk actually receive calls? and if so does it ring all 200 ports? |
| 19:31.59 | [j4jackj] | shamil: so buy a single channel |
| 19:32.06 | shamil | [j4jackj] I already have a single channel |
| 19:32.51 | shamil | Penguin, what is trunking defined as, per se |
| 19:33.05 | [j4jackj] | shamil: then you're done on the skype side |
| 19:33.07 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 19:33.41 | shamil | [j4jackj], it's the asterisk side where I'm failing |
| 19:33.47 | [j4jackj] | Oh. |
| 19:33.50 | [TK]D-Fender | shamil: Go register to them, go set up your peer. go place calls |
| 19:33.53 | [j4jackj] | Do they provide settings? |
| 19:33.54 | ChannelZ | ARGH fucking auto-playing video ads on websites piss me off |
| 19:34.09 | [j4jackj] | ChannelZ: srew that website all the way to hell |
| 19:34.14 | NoSu | how do you determine what version your svn build was around when compiled? |
| 19:34.18 | shamil | [j4jackj], they provide basic settings; the address, username, password and port |
| 19:34.27 | NoSu | Asterisk SVN-trunk-r330109M |
| 19:34.35 | [j4jackj] | shamil: I'll make a pastebin |
| 19:34.40 | shamil | [TK]D-Fender, setting up the peer.. I can't get it to register |
| 19:34.49 | shamil | tv [j4jackj] |
| 19:34.50 | shamil | tx* |
| 19:34.59 | [TK]D-Fender | shamil: You might want to consider showing us the debug then.... |
| 19:35.15 | ChannelZ | Synx|hm: There is no SIP trunk. |
| 19:35.34 | Penguin | Odd. I JUST got done covering that. |
| 19:35.56 | Synx|hm | understood and clear... i wish my customers would realize that |
| 19:36.13 | ChannelZ | Think of it more just like a set of authentication credentials. You can HAVE 200 independent users with different passwords, or have 1 and have 200 calls happening with it, all to different places. |
| 19:36.13 | Synx|hm | and i wish the major players like Avaya would realize that too!!! |
| 19:36.25 | [j4jackj] | http://sprunge.us/jQSZ |
| 19:36.28 | [j4jackj] | shamil: |
| 19:37.06 | Penguin | I think when someone says SIP trunk, they are talking about a group of channels, although completely and totally independent of each other, which all go between the same two peers. |
| 19:37.23 | ChannelZ | An ITSP might restrict the number of channels you can use (you pay for more) but otherwise it's not a finite thing. |
| 19:37.24 | [TK]D-Fender | So many things wrong with that sample.... |
| 19:37.32 | igcewieling | Penguin: In my experience a "sip trunk" is a peer. |
| 19:37.34 | Penguin | But they are really nothing more than a bunch of channels. There is no trunk. |
| 19:37.44 | shamil | [TK]D-Fender / [j4jackj]: debug log: http://pastebin.com/qSYEnp1y |
| 19:37.50 | shamil | let me grab my sip.conf |
| 19:37.53 | Synx|hm | in the context of my conversation with customers, i think they are doing exactly what ChannelZ said, they dont want to configure 500+ individual peers, just one and let the end device place calls out it as many concurrent times as it needs |
| 19:38.05 | Penguin | There's a huge difference between "trunk" and "peer." The terms should never be used interchangeably. |
| 19:38.09 | igcewieling | But since the world is full of idiots who insist on using the terms "SIP trunk" don't bother to try correcting them anymore. |
| 19:38.16 | igcewieling | I let them stew in their ignorance. |
| 19:38.35 | igcewieling | Penguin: right. A trunk is a PSTN term. |
| 19:38.56 | igcewieling | But that is not how people use the term trunk, they use it to mean asterisk peer |
| 19:39.08 | Penguin | and an IAX2 term. and an Ethernet term. and an elephant term. |
| 19:39.21 | shamil | [TK]D-Fender / [j4jackj]: http://pastebin.com/FA9rPhnF current sip.conf config |
| 19:39.23 | Synx|hm | well im glad i asked here then because i was really f'ing confused by its use in the IP world, it logically makes no f'ing sense in terms of IP |
| 19:39.26 | Penguin | If they are saying trunk when they mean asterisk peer, they're stupid. |
| 19:39.34 | ChannelZ | Well you can call it a SIP Trunk, I don't care, but the point is to stop thinking of it like a trunk. |
| 19:39.36 | [j4jackj] | Penguin: right on |
| 19:39.51 | igcewieling | Synx|hm: you are missing [general] |
| 19:40.03 | Penguin | Configure a PEER in sip.conf. That PEER entry is not a trunk. It's a peer. |
| 19:40.19 | Synx|hm | shamil: see igcewieling comment above its directed at you |
| 19:40.20 | [TK]D-Fender | shamil: All of your dilaplan for that has errors in it |
| 19:40.33 | igcewieling | Synx|hm: indeed it was |
| 19:40.37 | Penguin | Now the group of channels that the peer may create _COULD_ be thought of as a trunk, although not actually a trunk. |
| 19:40.43 | boom^time | if walks\quacks like a trunk |
| 19:40.44 | Penguin | But that's the group of channels, not the peer. |
| 19:41.00 | ChannelZ | I vote we name it a SIP Hose |
| 19:41.03 | [TK]D-Fender | shamil: As for the debug, I'm not seeing answers make it back |
| 19:41.04 | Synx|hm | igcewieling: just wanted to que shamil's notifcation system incase they were not tracking other conversations |
| 19:41.21 | [TK]D-Fender | shamil: Verify your WAN IP, and firewall/port forwardings to your server |
| 19:41.22 | Penguin | s/peer/hose/ |
| 19:42.14 | ChannelZ | See, much more fun. |
| 19:42.38 | jmetro | I cant remember if this works |
| 19:42.40 | igcewieling | Based on shamil's dialplan, it looks like he needs to learn Asterisk. |
| 19:42.45 | jmetro | multiple phones registered to one peer, yes/no? |
| 19:42.47 | shamil | [TK]D-Fender, NAT is an issue |
| 19:42.51 | Penguin | I think I will write a perl irssi script to always autoreplace the word peer with the word hose. |
| 19:42.55 | Penguin | |
| 19:42.55 | igcewieling | AND stop reading old documentation |
| 19:43.02 | jmetro | |
| 19:43.03 | ChannelZ | jmetro: no at the moment |
| 19:43.12 | igcewieling | jmetro: NEVER EVER |
| 19:43.24 | ChannelZ | jmetro: but * 12 can probably do it |
| 19:43.26 | jmetro | ChannelZ: ok, i couldnt remember. i had it in my head that it would function like a workgroup |
| 19:43.39 | ChannelZ | nope, currently asterisk only knows 1 IP per peer |
| 19:43.49 | shamil | igcewieling, I need to learn asterisk |
| 19:43.58 | igcewieling | jmetro: the last device to register is the one who wins |
| 19:44.07 | jmetro | igcewieling: yep, i remember that part. |
| 19:44.29 | igcewieling | shamil: Read the Asterisk book, "core show applications", "core show functions", UPGRADE*.txt and sip.conf.sample and extensions.conf.sample |
| 19:44.29 | shamil | [TK]D-Fender, it doesn't look like it's answering back |
| 19:44.33 | jmetro | shamil: read the book, read the book, read the MFin book |
| 19:44.52 | [TK]D-Fender | shamil: Contact: <sip:username@88.105.153.131:5060> <- is that a proper WAN IP to your server? Forwarding in place? Checked your firewalls? |
| 19:45.00 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 19:45.00 | [TK]D-Fender | shamil: it isn't... I've already told you that |
| 19:45.08 | igcewieling | This is pretty funny actually: Dial(SIP/101,30,t,r) so is voicemail(101|u) |
| 19:45.21 | [TK]D-Fender | igcewieling: As I said, the dialplan was wrong.. all of it |
| 19:45.35 | shamil | lemme verify the port forwarding is working |
| 19:45.47 | shamil | igcewieling, jmetro I've started to read it, haven't got very far, but I'm learning |
| 19:46.05 | igcewieling | shamil: the .sample config files will provide examples of specific things |
| 19:46.35 | shamil | thanks igcewieling :) |
| 19:46.38 | igcewieling | the UPGRADE*.txt files will tell you what things have changed so you can see when you are reading old documentation |
| 19:46.46 | Penguin | You don't necessarily need to read the entire book before starting with asterisk, but you really should read at least the sections pertaining to what you're working on. |
| 19:46.57 | igcewieling | "core show application dial" will show you the specifics of the Dial command |
| 19:46.57 | igcewieling | \ |
| 19:47.08 | Penguin | If you are going to configure sip devices, read that section. If you're ready to do dial plan, read that section. |
| 19:47.24 | Penguin | If you have time to read it all, read it all. |
| 19:48.02 | igcewieling | Buy a physical copy of the book. Put it in the bathroom. Leave your smartphone, read the book. |
| 19:48.30 | shamil | I'm waiting on a physical copy of the book |
| 19:49.07 | shamil | aha, apparently no one can see port 5060 from the wan |
| 19:49.34 | Penguin | That's pretty common. People forget to forward ports all the time. |
| 19:50.09 | ChannelZ | waits for the follow-up one-way audio question |
| 19:51.16 | *** join/#asterisk dgeary2 (~debian@pa49-187-94-200.pa.nsw.optusnet.com.au) |
| 19:52.04 | shamil | my routing device says that port forwarding is enabled |
| 19:52.33 | *** join/#asterisk protocoldoug (~doug@unaffiliated/protocoldoug) |
| 19:52.37 | shamil | I can't seem to telnet in on 5060 from the lan |
| 19:52.43 | Penguin | Of course not. |
| 19:52.49 | Penguin | SIP isn't TCP. |
| 19:52.56 | [j4jackj] | SIP is generally UDP |
| 19:53.04 | *** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
| 19:53.20 | Penguin | Unless your telnet client supports UDP, you won't be able to test that. |
| 19:53.35 | [j4jackj] | shamil: use a softphone |
| 19:53.36 | [j4jackj] | :D |
| 19:53.37 | Penguin | I use netcat with udp mode or a telnet that does have udp. |
| 19:53.58 | igcewieling | suspects he forwarded 5060/TCP |
| 19:54.02 | Penguin | heh |
| 19:54.46 | shamil | It was set to all, I've made it UDP specific now |
| 19:54.51 | [j4jackj] | heh |
| 19:55.34 | ChannelZ | BURRITO TIME! |
| 19:56.02 | Penguin | I hope it's a beef burrito. |
| 19:56.58 | ChannelZ | Steak, maybe. I don't like it rare though so I have to look at it first before I order. |
| 19:57.03 | *** join/#asterisk x05h (~x05h@66.55.141.76) |
| 19:58.03 | shamil | j4j, my hardphone is already registered :P |
| 19:58.22 | shamil | I know the irony :x |
| 19:59.24 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 19:59.38 | [TK]D-Fender | shamilj4j, my hardphone is already registered <- then * will be firghting with it |
| 19:59.50 | NoSu | how do you determine what version your svn build was around when compiled (Asterisk SVN-trunk-r330109M) |
| 20:00.01 | shamil | [TK]D-Fender ok |
| 20:00.15 | igcewieling | I doubt anyone here knows. |
| 20:00.37 | [TK]D-Fender | http://lists.digium.com/pipermail/asterisk-commits/2011-July/050523.html |
| 20:00.39 | Qwell | NoSu: No such comparison can be made. |
| 20:00.53 | [TK]D-Fender | another 5 seconds worth of google |
| 20:00.58 | Qwell | Releases are not made from trunk. |
| 20:01.19 | Penguin | The SVN trunk doesn't ever correspond with a release version? |
| 20:01.29 | mjordan | Nope |
| 20:01.31 | Qwell | Penguin: never |
| 20:01.41 | Penguin | I'm not sure what good it is, then. I don't understand the disconnection. |
| 20:01.52 | mjordan | heh |
| 20:01.56 | mjordan | branches are made from trunk |
| 20:01.59 | Qwell | Penguin: What kind of fruit does a tree produce? |
| 20:02.03 | igcewieling | trunk isn't good for much |
| 20:02.12 | mjordan | Qwell: svn rm trunk it is |
| 20:02.14 | Penguin | Okay, so the question simply was too specific. |
| 20:02.16 | Qwell | mjordan: DONE |
| 20:02.31 | Qwell | mjordan: speaking of being able to rm -rf trunk. We should talk at some point, still. |
| 20:02.39 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Software+Configuration+Management+Policies |
| 20:02.45 | NoSu | so basically if a plugin says its compatible with say 1.8 was the current stable build theres a chance it wouldnt work with the trunk |
| 20:02.48 | mjordan | Qwell: alrightly. Taco truck? |
| 20:03.01 | Qwell | mjordan: I was referring to cert stuffs. |
| 20:03.10 | Qwell | but, we should totally taco truck at some point too :p |
| 20:03.20 | mjordan | ah. Hm. I thought that was done. I shall put that on my queue. |
| 20:03.25 | igcewieling | NoSu: if a plugin wants 1.8 then install 1.8 |
| 20:03.41 | [TK]D-Fender | "trunk" is not a specific version |
| 20:03.46 | [TK]D-Fender | Trunk 5 years ago was not 1.8 |
| 20:03.46 | Qwell | mjordan: dunno, it may have been. I've never actually checked. |
| 20:04.16 | Qwell | (it wasn't - just checked) |
| 20:04.49 | shamil | I should probably note that I'm using elastix, but haven't touched the gui yet |
| 20:06.25 | igcewieling | puts shamil on /ingore |
| 20:07.17 | [TK]D-Fender | circumventing the GUI is a bad idea unless you know what you're doing |
| 20:07.39 | [TK]D-Fender | Which based on code & other samples does not appear too promising |
| 20:11.14 | navaismo | hahaha igcewieling |
| 20:12.07 | j4j | Should Digium buy Skype up? |
| 20:12.10 | [TK]D-Fender | The hardest part of goring a newb is getting the bull to focus on the target... |
| 20:12.25 | [TK]D-Fender | j4j: MICROSOFT owns Skype |
| 20:12.38 | j4j | I knoew |
| 20:12.40 | [TK]D-Fender | j4j: Your "relative fish size" radar is off |
| 20:13.09 | j4j | Microsoft is going to find Skype. unproftable sometime soon |
| 20:13.12 | _Corey_ | The Skype division at MSFT is about 100 times the size of Digium. |
| 20:13.40 | j4j | So maybe Digium should attempt to buy skype in just 6000 easy payments of $.05bn |
| 20:13.43 | j4j | :D |
| 20:13.46 | j4j | or sth |
| 20:13.58 | _Corey_ | ;) |
| 20:15.02 | j4j | It's sure as hell possible, and once Asterisk owns Skype, Skype servers (which are currenly Linux boxes at MSFT oh the irony) will be free for all purposes. |
| 20:15.50 | _Corey_ | ...and what would Digium's incentive to execute such a deal be? |
| 20:15.56 | Qwell | _Corey_: to own Skype, duh |
| 20:16.04 | igcewieling | j4j: It is helpful to occasionally contribute something USEFUL to the channe. |
| 20:16.17 | j4j | heh |
| 20:16.30 | j4j | It's general VoIP discussin, so somewhat on topic eh? |
| 20:16.34 | _Corey_ | smacks forehead |
| 20:16.34 | igcewieling | _Corey_: If Digium bought Skype I'd think Digium's intention would be to go bankrupt |
| 20:17.28 | j4j | But if Dgium bought Skype they would be buying a cash cow of Microsoft. and sure enough, they would add 'Standards mode' to Skype where it registers to Skype's new SIP servers... |
| 20:17.33 | igcewieling | j4j: not really. |
| 20:17.38 | j4j | It's all possible= |
| 20:17.54 | j4j | It's all possible... |
| 20:18.00 | [TK]D-Fender | Yes and pointless conjecture |
| 20:18.13 | j4j | Not pointless,... |
| 20:18.17 | igcewieling | j4j: please site your source that Skype is very profitable. |
| 20:18.28 | igcewieling | j4j: yes, pointless |
| 20:18.37 | j4j | Not as much as you think.... |
| 20:18.53 | j4j | There is no more source because Microsoft hid the facts |
| 20:18.55 | j4j | :D |
| 20:19.02 | Qwell | Cite your source that Skype has done anything but lose its owners billions of dollars? |
| 20:19.11 | igcewieling | The US congress suddenly all going sane is just as likely. |
| 20:19.16 | Qwell | it damn near did eBay in. |
| 20:19.53 | jmetro | US is doing some very scary anti-privacy =( |
| 20:19.56 | igcewieling | Qwell: you know j4j seldom makes any sense. |
| 20:20.09 | _Corey_ | Qwell: The original guys sold it to eBay and then bought it back and sold it to MSFT did fairly well... ;) |
| 20:20.34 | igcewieling | _Corey_: BT Barnum also did very well. |
| 20:20.40 | Qwell | _Corey_: This is true |
| 20:20.52 | [TK]D-Fender | checkout time, heading home... |
| 20:21.02 | igcewieling | "This way to the Egress" |
| 20:21.45 | j4j | I should talk to MSFT about splitting off the Skype division to me... Should I not? |
| 20:21.55 | igcewieling | j4j: Yes, you should. |
| 20:21.59 | _Corey_ | j4j: Splendid idea |
| 20:22.04 | igcewieling | Please, go do that now. |
| 20:22.15 | j4j | I'll send Sales an email :D |
| 20:22.38 | Penguin | I bet the only response you get out of it will be an autoresponse saying they got your email. |
| 20:26.23 | j4j | Heh, I think they've abandoned Redmond |
| 20:27.56 | WIMPy | Is there anything special about using ! patterns with realtime? Do they need to be escaped? |
| 20:31.50 | *** join/#asterisk Snipper_ (~IceChat9@mail.aytycrm.com.br) |
| 20:34.39 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 20:35.03 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 20:43.27 | j4j | WIMPy: I'm weird |
| 20:49.44 | *** join/#asterisk dash_ (~d45h@unaffiliated/dash-/x-7576607) |
| 20:50.32 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:53.34 | j4jackj | [TK]D-Fender: I think by giving you a number that gave dead air I cried stinking fish. |
| 20:58.12 | ChannelZ | belches |
| 20:58.16 | Penguin | Well, there's always that. |
| 21:05.01 | WIMPy | j4jackj: that's understood. |
| 21:05.33 | j4jackj | WIMPy: well I think now this DID will work |
| 21:07.42 | Penguin | The more important question is why didn't it work before. |
| 21:08.13 | j4jackj | Because my dialplan was crewed |
| 21:08.16 | j4jackj | *screwed |
| 21:08.28 | WIMPy | Noone into realtime dialplan stuff and patterns? It's not realtime any more now, but I'd like to know why it didn't work with realtime. |
| 21:08.49 | WIMPy | . o O ( if it was only the dialplan... ) |
| 21:09.00 | j4jackj | Because it was too hard to implement without needing Gamin. |
| 21:09.24 | j4jackj | . o O ( If I could get interested people to call my DID ) |
| 21:09.55 | Penguin | pops some thought-bubbles |
| 21:09.58 | igcewieling | WIMPy: only crazy people use realtime. Apparently they are all AFK |
| 21:09.59 | WIMPy | j4jackj: do what everyone does: Put it up on some dating site. |
| 21:10.11 | j4jackj | Nah |
| 21:10.20 | j4jackj | Maybe my blog, but not a datingsite |
| 21:10.23 | cusco | what's wrong with realtime? |
| 21:10.30 | WIMPy | igcewieling: I guess so. |
| 21:10.31 | cusco | ah dialplan |
| 21:10.32 | cusco | no |
| 21:10.40 | cusco | we use realtime peers and queues |
| 21:10.58 | igcewieling | cusco: issue a sip reload, then "sip show peers" |
| 21:11.13 | igcewieling | they will all be gone until there is traffic from them/to them |
| 21:11.29 | cusco | true.. |
| 21:11.35 | cusco | how is that bad? |
| 21:11.45 | igcewieling | you also can't seperate out the write ops of realtime from the read ops, so using a local replicated database for realtime is totally pointless |
| 21:12.00 | igcewieling | cusco: you have to manually load any peers which are offline |
| 21:12.09 | igcewieling | qualify doesn't work until the peer generates traffic |
| 21:12.40 | cusco | igcewieling: manually load? er.. they are queue members, queue will try to ring them via Local/channels |
| 21:12.54 | igcewieling | cusco: I'm referring to sip peers |
| 21:12.56 | cusco | and dialplan will dial SIP/peer and they will become available |
| 21:13.11 | igcewieling | cusco: but UNTIL you dial, the peer won't even exist. |
| 21:13.29 | cusco | how can that be bad? (i'm not saying it isn't) |
| 21:13.32 | igcewieling | sip show peers will not show it, for example. |
| 21:13.50 | igcewieling | cusco: It is incredibly annoying when troubleshooting |
| 21:13.58 | cusco | I can just query the database orderying by regseconds |
| 21:14.08 | igcewieling | if the peer not there because it is not there or is it not there because it has not registered yet |
| 21:14.19 | cusco | I see... |
| 21:14.32 | cusco | well its usefoul to integrate other apps with asterisk stuff |
| 21:14.46 | cusco | I do have a problem with realtime |
| 21:15.00 | igcewieling | cusco: the read ops .vs. write ops is my main problem with it. |
| 21:15.15 | cusco | queue doesn't know the peer state until I do queue show <queue> |
| 21:15.22 | cusco | :( |
| 21:15.24 | igcewieling | I STILL have not gotten voicemail greetings to be stored in the database. |
| 21:15.31 | igcewieling | cusco: I care nothing about queues. 8-) |
| 21:16.06 | WIMPy | because you always cut in line? |
| 21:17.10 | cusco | hehehe |
| 21:32.06 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.14) |
| 21:34.43 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 21:34.43 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 21:50.44 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 22:01.35 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 22:07.59 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.75) |
| 22:08.46 | j4jackj | hi pabelanger |
| 22:11.14 | pabelanger | j4jackj, hello |
| 22:11.58 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 22:12.14 | j4jackj | Hello all |
| 22:12.32 | j4jackj | I wonder why I have set up a PBX conference service..? |
| 22:13.09 | Penguin | Are you asking us if you wonder why? |
| 22:14.29 | j4jackj | No I'm asking why I set it up? |
| 22:14.53 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 22:14.56 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 22:15.41 | j4jackj | Worse, it's PSTN accessible. |
| 22:16.03 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 22:16.11 | WIMPy | Yes, that's bad. |
| 22:16.11 | j4jackj | Hi Chainsaw |
| 22:16.18 | Chainsaw | Hello. |
| 22:16.52 | j4jackj | I use Asterisk because I can use it for teleconferencing, no one has signed up yet, but apb1963 I think has. |
| 22:17.36 | ChannelZ | What service are you providing that we all can't already do? |
| 22:18.20 | j4jackj | Not al ot |
| 22:18.38 | j4jackj | Just 'it's set up by someone who will make sure things don't crossplode' |
| 22:18.43 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 22:19.54 | ChannelZ | except it hasn't worked yet has it |
| 22:20.09 | *** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com) |
| 22:21.57 | *** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net) |
| 22:30.16 | j4jackj | heh |
| 22:30.27 | j4jackj | then agan no one has used it |
| 22:32.57 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 22:34.48 | *** join/#asterisk ziegleka (~ziegleka@keeper.etnetera.cz) |
| 22:35.08 | apb1963 | what? my ears are burning |
| 22:35.47 | WIMPy | From professional filth? |
| 22:36.05 | apb1963 | ??? |
| 22:36.49 | apb1963 | No, I picked up a hot iron when the phone rang. |
| 22:37.28 | ChannelZ | That's an old Helen Keller joke |
| 22:39.37 | j4jackj | heehe |
| 22:40.05 | j4jackj | apb1963: you remember signing up for Dial 7 ? |
| 22:40.34 | igcewieling | interesting seeing only one side of a conversation.... |
| 22:40.35 | apb1963 | nope |
| 22:40.48 | apb1963 | which side igcewieling? |
| 22:41.00 | igcewieling | apb1963: your side |
| 22:41.14 | igcewieling | and ChannelZ |
| 22:41.19 | apb1963 | well I'm just glad you're on my side |
| 22:41.41 | j4jackj | apb1963: I believe you were the one to dial 71@99.199.11.127 were you not? |
| 22:42.12 | apb1963 | j4jackj: I'm not admitting to anything that might incriminate me |
| 22:42.36 | j4jackj | apb1963: This is not incriminating. I'm just saying I believe you dialled that magic number. |
| 22:42.39 | j4jackj | :D |
| 22:43.01 | igcewieling | Heh, customer on hosted service moved to an on-site server, now they are moving from VoIP to PRI |
| 22:43.13 | j4jackj | heh |
| 22:43.17 | j4jackj | Weird eh? |
| 22:43.29 | j4jackj | Why, I ask are they moving to PRI? |
| 22:43.42 | WIMPy | Oh, it has begun? |
| 22:44.08 | j4jackj | What has? |
| 22:44.52 | WIMPy | The comeback of common sense. |
| 22:45.32 | j4jackj | Commonsense to what? |
| 22:45.42 | j4jackj | As in to use PRI rather than VOIP? |
| 22:46.08 | WIMPy | yes |
| 22:46.19 | j4jackj | That's silly. |
| 22:46.24 | igcewieling | WIMPy: I doubt it. |
| 22:46.30 | j4jackj | VoIP and PRI should be combined. |
| 22:46.49 | WIMPy | Why? |
| 22:47.21 | igcewieling | WIMPy: Their current location (temporary after Hurricane Sandy) loses power 3 - 4 times per week for several hours. |
| 22:47.57 | igcewieling | WIMPy: the ONLY Logical thing I've seen them do is order the PRI 8-| |
| 22:52.07 | j4jackj | How is PRI more logical than VoIP |
| 22:52.38 | *** join/#asterisk russum (~russum@ool-4a599ffe.dyn.optonline.net) |
| 22:52.48 | WIMPy | Milk is logical. |
| 22:53.10 | apb1963 | not if you're lactose intolerant |
| 22:53.13 | [TK]D-Fender | Biscuits |
| 22:53.29 | apb1963 | British or American? |
| 22:53.42 | j4jackj | HuH? |
| 22:53.49 | *** part/#asterisk russum (~russum@ool-4a599ffe.dyn.optonline.net) |
| 22:53.59 | apb1963 | Biscuits are different depending on what country |
| 22:54.08 | j4jackj | yup |
| 22:54.24 | j4jackj | british biscuits are american cookies. it sucks. |
| 22:54.25 | apb1963 | But milk intolerance isn't |
| 22:54.45 | apb1963 | Yeah, because our biscuits are yummy |
| 22:54.57 | apb1963 | especially with a little butter |
| 22:55.05 | j4jackj | APB1963 are you lactse intolerant? |
| 22:55.14 | apb1963 | j4jackj: Why do you ask? |
| 22:55.17 | Penguin | Why puts butter on cookies? |
| 22:55.22 | j4jackj | Because you mention it |
| 22:55.32 | j4jackj | Penguin: IDK |
| 22:55.37 | apb1963 | Penguin: Because it's yummy and the only way to eat a buscuit |
| 22:55.47 | j4jackj | heh |
| 22:55.53 | apb1963 | s/buscuit/biscuit/ |
| 22:56.04 | j4jackj | apb1963: are you a brit who has lived in the USA for 18 years? |
| 22:56.11 | j4jackj | abc |
| 22:56.15 | j4jackj | s/abc/def/ |
| 22:56.19 | j4jackj | :D |
| 22:56.22 | apb1963 | j4jackj: No I'm an American that has always lived in America |
| 22:56.27 | j4jackj | apb1963: heehe |
| 22:57.00 | Penguin | Cookies, I said. |
| 22:57.22 | j4jackj | It's probably a Britticism. |
| 22:57.24 | j4jackj | :D |
| 22:57.46 | Penguin | I've never known anyone to butter a cookie. |
| 22:57.48 | apb1963 | I prefer oatmeal raisin cookies but will settle for chocolate chip |
| 22:58.04 | apb1963 | I don't butter cookies... I butter biscuits |
| 22:58.11 | j4jackj | I prefer oatmeal chocolate |
| 22:58.19 | apb1963 | that's just weird |
| 22:58.22 | j4jackj | Insanity |
| 22:58.23 | Penguin | That's a new one. |
| 22:58.37 | j4jackj | .. .. .. . ... .. . . .. . |
| 22:58.38 | apb1963 | He made it up... it doesn't exist |
| 22:58.43 | j4jackj | Yup :D |
| 22:59.29 | j4jackj | apb1963: shall we take this to the audioconference? It's not acceptable for the channel |
| 22:59.59 | apb1963 | no but thanks for the offer :) |
| 23:02.00 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
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| 23:57.32 | ChannelZ | Your system is back to handing out its LAN IP again. |