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| 00:37.19 | leifmadsen | omg I just sent a fax via T.38 |
| 00:37.31 | leifmadsen | all the planets must be aligned |
| 00:37.34 | navaismo | XD |
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| 00:55.02 | pabelanger | leifmadsen, heh, I've been working with T.30 over a satellite recently. Not fun |
| 00:55.21 | pabelanger | wish we had t.38 |
| 01:00.55 | coppice | leifmadsen: if you send T.38 via any of the planets you'll get protocol timeouts. even moon bouncing is pushing it |
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| 01:10.47 | leifmadsen | coppice: :) |
| 01:10.54 | leifmadsen | pabelanger: btw, i filed asterisk-21828 today |
| 01:10.54 | LieutPants | [ASTERISK-21828] [Status: Triage] [patch] app_meetme.so hints load as Unavailable instead of Idle on start up - https://issues.asterisk.org/jira/browse/ASTERISK-21828 |
| 01:11.03 | leifmadsen | pabelanger: matching reviewboard request to boot |
| 01:11.08 | leifmadsen | just for you |
| 01:13.01 | tzanger | coppice: bouncing fax transmissions off the moon just to say you did it is about the only real use a fax should have these days |
| 01:13.27 | coppice | tzanger: don't start that crap |
| 01:13.54 | tzanger | faxes (should) have real utility these days? |
| 01:14.27 | coppice | yes, because nobody has introduced something that would genuinely make them obsolete |
| 01:14.58 | coppice | its not enough for a better approach to be possible. it has to exist and be widely deployed |
| 01:15.24 | tzanger | I suppose. I mean I *know* they're used still but man, it's like the DOS APIs in Windows.. just can't shake them |
| 01:16.18 | tzanger | scan&email is pretty damned widespread, tacking on a receipt response seems to pretty much nail it although that little detail is a bit of a gotcha |
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| 01:18.09 | coppice | how often to you send or receive scan and email messages? emails with camera phone pictures attached are more of a real world FAX replacement than that |
| 01:18.10 | tzanger | at any rate I certainly wasn't trying to make light of all the work you've done not only in signal processing for faxing but the entire faxing implementation. As I said faxing is a tech that we should have been able to shake by now but just can't |
| 01:19.39 | coppice | 90% of the people complaining about FAX being obsolete seem to be people who just can't get a VoIP system to work properly :-) |
| 01:19.48 | tzanger | coppice: of signed contracts? often enough and widespread enough for my (small) data set that faxing isn't necessary. faxing camera phone images is something I haven't yet had to do |
| 01:19.53 | tzanger | heh |
| 01:20.55 | tzanger | nah my little linksys spa-whatever box works great for faxing when I need it. my * box works well enough |
| 01:22.24 | phix | coppice: nope, I have FAX setup and it is obsolete :) |
| 01:22.37 | phix | although fax to email gateways are the way to go |
| 01:23.42 | phix | laweyers / solicitors <3 faxes as they can charge a shit load per page to send or receive them |
| 01:23.53 | coppice | most people don't like fax to email gateways. they expect FAX to be direct and immediate. T.37 would have been the way to go if gateways had been more acceptable |
| 01:24.31 | coppice | lawyers, doctors, real estate people, and accounts depts worried about sarbanes oxley seem to be the volume users now |
| 01:24.47 | carrar | mmm fax |
| 01:25.02 | phix | coppice: yup, fax to email with pdf attachment of the fax, then just setup something in the background to download it, andprint it off automatically if needed, instant! |
| 01:25.02 | carrar | I have two fax machines next to my desk |
| 01:25.09 | phix | plus you also have an electronic copy too |
| 01:26.01 | tzanger | it's odd that sarbanes-oxley is the driving force, centralized email systems like exchange are compliant aren't they? |
| 01:26.41 | carrar | everyone in Japan has fax machines too |
| 01:27.41 | coppice | tzanger: I don't know the details of why they drives FAX. you'd think with the volumes of material passing along supply chains that they'd would have established a legally OK alternative |
| 01:30.49 | phix | exchange is not compliant to any standard I thought |
| 01:30.56 | phix | not even the ones designed by MS |
| 01:31.54 | tzanger | phix: :-) |
| 01:32.51 | tzanger | a quick google search suggests exhcnage server can be made compliant. probably means that it's cheaper to just fax and store the paper |
| 01:33.48 | coppice | "can be made complaint" often means with years of effort and millions of dollars. its like "FAX can be made obsolete" |
| 01:33.56 | tzanger | :-) |
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| 01:36.55 | saint_ | i have to setup a fax extension in asterisk, and it sux ! (to use fax) |
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| 01:39.21 | phix | saint_: yeah but it can be done :) I have done it a few times |
| 01:39.27 | phix | with hylafax |
| 01:39.55 | tzanger | I've used it a number of times, both receiving electronically as well as straight fax passthrough |
| 01:40.26 | phix | Havn't it done it over T.38 yet though as my provider doesn't support it |
| 01:41.02 | phix | well they support it but they don't guarentee they will route your connection to another host that supports it |
| 01:49.38 | saint_ | phix: oh yeah, I have it working (receiving at least) |
| 01:49.52 | saint_ | phix: over sip, using voip.ms |
| 01:50.04 | saint_ | works with google voice too, but not the detection |
| 01:52.23 | phix | yay |
| 01:52.38 | igcewieling | faxing going away is a great goal, like world peace, and equally as practical |
| 01:52.57 | phix | yeah detection is a pain, on my system (with a PSTN line) I just wait 5 secs for the fax detection to kick in :P |
| 02:15.25 | apb1963 | coppice: The legally OK alternative is called EDI and it's been in place for over a decade. http://en.wikipedia.org/wiki/Electronic_data_interchange |
| 02:22.24 | coppice | apb1963: if that were truly the answer the accounts depts of all these multinationals wouldn't have a row of FAX machines |
| 02:24.08 | coppice | bizarrely, when I see external auditors working in multinationals, they are sifting through mountains of paperwork, rather than electronic records |
| 02:24.47 | apb1963 | that's not my fault |
| 02:25.35 | coppice | EDI is really old. People used to exchange EDI records using the file transfer mode of the FAX protocol back in the 90s |
| 02:25.37 | tzanger | it's coppice's... if his software didn't work right they'd have had to have moved on :-) |
| 02:25.52 | apb1963 | You can't MAKE people change - well you can, but you have to be an 800 pound gorilla or bigger to do so. |
| 02:26.32 | apb1963 | Like I said.. it's been in place for over a decade. |
| 02:26.37 | coppice | there are often some devious iffy things going on with accountants, which make them act in none obvious ways for good reasons |
| 02:26.53 | apb1963 | I'm guessing it's their lack of technical prowess |
| 02:27.23 | apb1963 | they can do simple math... but that's about it. |
| 02:27.31 | coppice | hardly. the have the cash to pay for pretty much anything they want. |
| 02:27.45 | coppice | and they use EDI heavily. the just use FAX in parallel |
| 02:27.50 | apb1963 | why would someone WANT something they don't understand? |
| 02:28.19 | coppice | people understand only a tiny fraction of the stuff they work with. that's normal these days |
| 02:28.27 | apb1963 | ah. so they do use EDI. |
| 02:28.43 | coppice | of course. |
| 02:29.14 | apb1963 | People need paper. It's too easy to lose electrons. |
| 02:29.55 | coppice | that's true, although much of the printing we do these days, especially laser printing, can lift off the page after several years |
| 02:30.30 | apb1963 | where several = decades |
| 02:30.46 | robl^ | what gets me is that in my current employer's firm is they insist on fax.. yet our faxes are all electronic. just ends up as PDFs in their email. Why not just do email to start with? |
| 02:31.08 | apb1963 | It goes back to not understanding technology. |
| 02:31.30 | coppice | depends on the climate. if its warm and humid a stack of laser printed sheets can transfer chunks of text between the front of one sheet and the back of the next one in a few months |
| 02:31.47 | apb1963 | Familiarity. Not wanting to move out of their comfort zone. |
| 02:32.04 | apb1963 | That may be true. i live in a fairly dry climate. |
| 02:32.26 | coppice | its quite dry here today. its only about 90% humidity |
| 02:33.05 | apb1963 | it doesn't get that humid here when it rains. |
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| 02:33.35 | Free99 | hey guys. looks like infiltrated.net is down, I'd have liked to use the blacklist... any suggestions? |
| 02:34.00 | apb1963 | use the purplelist... much prettier :) |
| 02:34.44 | Free99 | apb1963, :P |
| 02:34.54 | Free99 | everyone knows black is cool |
| 02:35.01 | Free99 | except the cops apparently |
| 02:35.06 | Free99 | but anyway |
| 02:36.27 | Free99 | I've got some guests dialing into my system and trying to make calls to some weird numbers in what looks like israel... I have an AGI that authenticates guests, but I'd like to figure out how to block those ridiculous 1900 numbers etc that people made up in other countries |
| 02:36.45 | Free99 | is there such a list of "bad" numbers? |
| 02:37.13 | coppice | I guess most arid places look pretty much like israel |
| 02:37.21 | WIMPy | All numbers that aren't yours. |
| 02:38.34 | Free99 | WIMPy, but my actual users are trying to call family as they are supposed to... |
| 02:39.00 | Free99 | iunno, I'm just nervous that they may eventually break into my server or something |
| 02:39.30 | WIMPy | That's the idea. |
| 02:39.44 | WIMPy | And the ways they try can be very entertaining. |
| 02:40.30 | Free99 | WIMPy, do you think the upgrade from 1.8 to 11 is worth it just for the security logs? |
| 02:40.50 | Free99 | I'm just not too keen moving away from my distro's supplied packages |
| 02:43.23 | WIMPy | I would have said yes if you had not restricted it to the security logs. |
| 02:44.06 | Free99 | I mean 1.8 doesn't support that...right? |
| 02:45.23 | WIMPy | Don't think so. |
| 02:45.33 | WIMPy | But I haven't used them. |
| 02:46.32 | Free99 | you're not worried you'll get a $10000 phone bill one of these days? |
| 02:47.28 | WIMPy | Nope |
| 02:48.10 | WIMPy | I don't get my bill in dollars :-) |
| 02:49.04 | Free99 | (facepalm) lol |
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| 04:48.15 | robert_ | so is there a way I can do classes in the mysql config? |
| 04:54.14 | sean5699 | Hi, has anyone used the polycom 560 with asterisk? I'm wondering how well it works |
| 05:10.32 | [TK]D-Fender | seanWorks just fine |
| 05:29.47 | robert_ | can you store peers inside mysql? |
| 05:29.53 | robert_ | (or any database) |
| 05:31.29 | [TK]D-Fender | yes |
| 05:31.34 | [TK]D-Fender | ~book |
| 05:31.34 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 05:31.35 | [TK]D-Fender | ^^^ |
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| 05:58.15 | Rhomber | great, asterisk is still broken (with the sip registration bug that's suppose to be fixed) |
| 05:58.23 | Rhomber | though, it took longer to break |
| 05:58.34 | Rhomber | this is in 11.4.0 :( |
| 05:59.10 | Rhomber | after a while, none of the SIP clients can communicate with asterisk.. and nothing comes up in the logs |
| 05:59.39 | Rhomber | doing a "sip reload" immediately returns with no output, then when it's done again I get "Previous SIP reload not yet done" |
| 05:59.51 | Rhomber | just as I did with 11.2.1 |
| 06:20.19 | sean5699 | anyone ever use the polycom vvx 500 phone with asterisk? |
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| 06:29.22 | robert_ | I can't find anything to do with user classes.. does this mean you can't do them inside odbc config? |
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| 06:56.29 | robert_ | anybody? |
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| 08:19.10 | robert_ | weird |
| 08:20.38 | robert_ | "SQL Prepare failed![SELECT * FROM voicemessages WHERE dir=? AND msgnum=?]" |
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| 08:30.22 | fling | How to properly log calls to syslog? |
| 08:32.24 | fling | I have this in my logger config and calls are not logged > syslog.local0 => notice,warning,error |
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| 08:41.13 | fling | added verbose, now it is logging |
| 08:41.18 | fling | fling: thanks |
| 08:41.23 | fling | fling: yw |
| 08:41.39 | ChannelZ | well my job here is done |
| 08:46.27 | fling | ChannelZ: hello :D |
| 08:47.25 | ChannelZ | Ahoy |
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| 08:50.59 | fling | I have multiple sip clients. Each of them has some unique extensions and all of them has a lot of equal extensions in dialplan. |
| 08:51.29 | fling | I need to put each client into different context to be able to set different extensions to each of them, right? |
| 08:51.54 | fling | But I do not want to duplicate all the extensions :| |
| 08:52.42 | fling | May I somehow set extensions for multiple contexts? |
| 08:53.34 | ChannelZ | Well I don't understand the question. The extensions are either different or they are the same. |
| 08:54.03 | fling | ChannelZ: right, clients have both differnt and same extensions :P |
| 08:54.53 | ChannelZ | If there are shared extensions, you could put those into a 'shared' context and then include them from the 'unique' context to a given peer |
| 08:55.21 | fling | ChannelZ: include? hmm hmmm is it in example extensions.conf? |
| 08:55.32 | fling | looks like it is what I need, thanks |
| 08:55.37 | ChannelZ | include => context |
| 08:55.55 | ChannelZ | where context is the name of the context to include |
| 08:56.26 | fling | I'm about to simplify my setup. |
| 08:56.56 | ChannelZ | This is handy even in simple setups; For instance I have an [out-local] context for local calls, [out-ld] context for long distance, and [internal] for internal extensions |
| 08:58.33 | ChannelZ | via including in each context (out-ld includes out-local and internal, for instance) and setting a peer for a particular context, they can either make all kinds of calls, local and internal calls, or only internal. |
| 08:59.34 | fling | ChannelZ: smart! Is it documented somewhere? |
| 09:00.03 | ChannelZ | well include is, but how you implement it is up to your imagination and/or needs |
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| 09:07.39 | robert_ | how do I get asterisk to correctly authenticate extensions in mysql? |
| 09:09.55 | wdoekes | s/extensions/devices/ |
| 09:10.18 | robert_ | it keeps saying invalid password when I try to connect with a softphone |
| 09:10.51 | wdoekes | sip show peer PEERNAME load |
| 09:11.52 | robert_ | it's not finding any peers x.x |
| 09:17.09 | robert_ | I can do something like, 'realtime load sipusers username [username]' |
| 09:17.16 | robert_ | but not 'sip show peers' |
| 09:20.24 | *** join/#asterisk RiceCracker (~RiceCrack@59.152.236.158) |
| 09:20.52 | RiceCracker | Hello |
| 09:21.42 | RiceCracker | I have a stupid question but since there are no stupid questions only stupid people, i'm stupid and i have a question. |
| 09:22.45 | RiceCracker | If I have installed the G.729 codec and I make a call out over my T1 line, does that take up 1 codec? |
| 09:24.30 | wdoekes | robert_: how is asterisk supposed to know that your sipusers table holds sip friends? |
| 09:24.38 | RiceCracker | and if I make a call from my mobile phone to a conference on my PBX, will that take up a codec? |
| 09:24.41 | wdoekes | look at extconfig.conf |
| 09:25.21 | robert_ | I told it, or I think I told it at least |
| 09:25.41 | wdoekes | RiceCracker: you need 1 codec per concurrent transcoding session. so if both legs speak g729, you need 0. if one of them speaks g729, you need 1 |
| 09:25.46 | robert_ | 'sipusers -> odbc,asterisk,sip_users' |
| 09:25.51 | robert_ | er |
| 09:25.54 | robert_ | 'sipusers => odbc,asterisk,sip_users' |
| 09:26.18 | wdoekes | 1: sipusers does nothing, you need to set sippeers |
| 09:26.27 | wdoekes | 2: sip_users isn't the name of the table you just mentioned |
| 09:27.47 | RiceCracker | thanks wdoekes, so if i make a call from my pbx to my mobile, that will take up 1 g729 codec correct? |
| 09:28.20 | wdoekes | I have no idea. that depends on which codecs are preferred |
| 09:28.24 | wdoekes | try it! |
| 09:28.26 | robert_ | wdoekes: still does nothing. |
| 09:28.36 | RiceCracker | if my g729 is set to preffered |
| 09:29.08 | RiceCracker | i have g729 set to be used first. |
| 09:29.40 | robert_ | sip show peers does very much nothing |
| 09:30.09 | wdoekes | RiceCracker: if the other end doesn't do g729, you can prefer it all you want ;) |
| 09:30.29 | wdoekes | robert_: 3: I didn't say 'sip show peers' would |
| 09:30.44 | wdoekes | robert_: sip show peer XYZ load <-- note the "load" part |
| 09:31.00 | wdoekes | core set debug 20 |
| 09:31.03 | wdoekes | and then do the load |
| 09:31.18 | wdoekes | and then look at the debug log to see what queries were done |
| 09:31.58 | RiceCracker | reason is i'm trying to figure out how we ended up running out of codec licecense becuase it inturn cause a codec translation error when trying to switch from g729 to ulaw. what i'm trying to figure is how all licenses got used. so what i want to know is if making calls over our T1 which connects to |
| 09:32.17 | RiceCracker | a local pstn, will it use up a codec |
| 09:32.37 | RiceCracker | i.e. i call my mobile phone from my desk |
| 09:32.52 | wdoekes | can't you just try it? and see if your extra call takes up an extra license? |
| 09:33.11 | wdoekes | I have no idea which protocols/lines use which kind of codec, but you can easily see it when making the call |
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| 09:41.15 | RiceCracker | okay so apparently it doesn't |
| 09:41.49 | RiceCracker | i cant see any inuse when i go though the T1, but of course uses it when connecting between two boxes |
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| 09:46.53 | RiceCracker | thanks wdoekes. would this error be a programming error if i had all the pbx set to use g729 over ulaw, but the phone programming says the native default is ulaw? "chan_sip.c: Asked to transmit frame type g729, while native formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw)" |
| 09:47.10 | RiceCracker | because after that error i get this "Codec mismatch on channel SIP/5024-000003c8 setting write format to g729 from ulaw native formats 0x4 (ulaw)" |
| 09:47.27 | RiceCracker | followed by "Unable to find a codec translation path from 0x4 (ulaw) to 0x100 (g729)" |
| 09:47.54 | RiceCracker | my guess is i ran out of g729 so it tried to go back to ulaw but failed... |
| 09:52.17 | wdoekes | are you using local channels? |
| 09:53.50 | fling | May not I use skypiax? Where is it's source? |
| 09:54.56 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 09:54.57 | RiceCracker | wdoekes: call was made between two asterisk servers connected via IAX |
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| 09:58.44 | wdoekes | mm.. ok. I can't answer whether it's a programming issue or you were out of codecs. but the latter is probably more likely |
| 10:01.39 | RiceCracker | thanks wdoekes |
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| 10:15.49 | fling | Where may I get chan_skypiax.so? |
| 10:19.04 | RiceCracker | i think its dead |
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| 10:24.58 | fling | RiceCracker: do I need to use SipToSis instead? |
| 10:25.35 | fling | I want to have a simple setup. |
| 10:36.21 | ghost75 | is it normal that originate is not working when remote side is busy? |
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| 12:30.18 | sweeper | anyone know a ballpark estimate for concurrent meetme participants with no transcoding on an ec2 xl-highcpu instance? |
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| 12:33.00 | sweeper | or confbridge, not picky :) |
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| 13:21.52 | jmetro | what is the standardly accepted good Android softphone? |
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| 13:23.54 | [TK]D-Fender | jmetro: Google Nexus 4 (by LG) 16 GB |
| 13:24.05 | [TK]D-Fender | AKA "latest vanilla" |
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| 13:24.42 | jmetro | [TK]D-Fender: i meant the softphone app <.< |
| 13:25.02 | jmetro | And i think the HTC one has got the nexus 4 beat in terms of android smartphone |
| 13:25.18 | jmetro | Or the S4 vanilla. |
| 13:29.18 | igcewieling | sweeper: all MeetMe is transcoded for mixing |
| 13:29.49 | igcewieling | jmetro: all softphones suck |
| 13:30.36 | [TK]D-Fender | igcewieling: So is confbridge :) |
| 13:31.20 | [TK]D-Fender | jmetro: I've head good things about Bria on them, and I've used 3CX Phone lightly and was OK. |
| 13:32.25 | sweeper | igcewieling: soooo basically it's gonna take several linked conference bridges ? |
| 13:35.16 | igcewieling | sweeper: that is what most people do for conferences with large numbers of people. |
| 13:35.53 | [TK]D-Fender | sweeper: Rather than "how many can it survive" how about telling us what you NEED. |
| 13:36.34 | sweeper | [TK]D-Fender: I've asked for that information. right now the customer has requested "a bazillion" |
| 13:37.07 | sweeper | I answered asking if it's a metric bazillion or an imperial bazillion |
| 13:37.18 | [TK]D-Fender | Correct response... |
| 13:37.20 | sweeper | igcewieling: good to know |
| 13:39.01 | sweeper | any ideas on a safe approximation per EC2 instance? I mean if they come back with 500, I'd like to be able to tell them more or less how much it costs |
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| 14:16.10 | jacekowski | i've got question about echo cancellation using oslec and friends |
| 14:16.24 | jacekowski | does it do anything to outgoing voice? |
| 14:16.35 | jacekowski | (ISDN) |
| 14:17.10 | igcewieling | jacekowski: no. |
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| 14:17.45 | jacekowski | what else could cause horrible sound quality on calls from isdn to sip |
| 14:17.48 | igcewieling | echo needs to be canceled on the inbound direction where the audio is converted from TDM to SIP. |
| 14:17.55 | jacekowski | as in like crackling skipping type thing |
| 14:18.07 | MLNoah | Can parking hints be published via res_corosync or XMPP PubSub? e.g. park:171@parkinglot_testguy |
| 14:18.07 | igcewieling | jacekowski: that is not echo |
| 14:18.50 | jacekowski | i know it's not echo, i just though it may be echo canceller going into some weird state messing things up |
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| 14:22.44 | aberrios | what verbosity to I turn up to see why asterisk wont load a module? core set verbose and debug seem to have no effect |
| 14:23.14 | igcewieling | maybe your transmit levels are too high |
| 14:25.20 | igcewieling | aberrios: which module are you having trouble loading? |
| 14:25.26 | aberrios | chan_sip.sop |
| 14:25.37 | aberrios | works on another node with same versions..... |
| 14:25.44 | aberrios | very odd |
| 14:25.46 | [TK]D-Fender | aberrios: Show us your attempt to load it and go prove that sip.conf is sane |
| 14:26.23 | aberrios | ah sip.conf is empty.....so it shouldnt have loaded on node1....thanks |
| 14:26.29 | igcewieling | aberrios: if there is an error loading a module you should see the error regardless of verbose or debug level |
| 14:27.00 | aberrios | i got it now... thanks |
| 14:27.19 | aberrios | highfives [TK]D-Fender and igcewieling |
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| 14:28.54 | aberrios | desperately needs chef and cobbler up and going |
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| 14:35.33 | aberrios | . o O (consistent builds...consistent builds...consistent builds) |
| 14:35.59 | MLNoah | hm. is there a way to clear dynamic hints from the hint cache without restarting Asterisk? |
| 14:36.19 | Chainsaw | MLNoah: That'd be neat. I don't know of one. |
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| 14:47.01 | igcewieling | aberrios: that is what the .opts files are for |
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| 14:49.41 | aberrios | igcewieling, I'm just using packages atm |
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| 14:51.37 | aberrios | damn run out of caffeinated beverage |
| 15:00.14 | jmetro | sounds like you need Bawls |
| 15:02.25 | xcom | ma Bawls |
| 15:05.08 | jmetro | xcom: enemy unknown |
| 15:05.21 | xcom | lol |
| 15:05.27 | robert_ | igcewieling: hai |
| 15:05.42 | jmetro | =p |
| 15:07.14 | robert_ | igcewieling: btw, there's a "setenv" so we're fine on thast front. But now our users aren't coming up. |
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| 15:11.48 | igcewieling | robert_: with realtime peers don't exist in asterisk until they connect or register |
| 15:12.28 | robert_ | yeah, that's the problem. they're NOT registering. I keep trying, but all I get is invalid password. |
| 15:15.00 | igcewieling | do a sip show peer X load where x is one of your peers. then do a sip show peers and it should show up. If it doesn't, you have a problem with your Realtime setup |
| 15:16.03 | igcewieling | you can also enable query logging on your sql server and see what queries realtime is making against the DB. I had to use that a few months ago in order to figure out a realtime issue. |
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| 17:08.45 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 17:08.45 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), 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 |
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| 17:42.00 | raidghost | On a server where asterisk is running. is there any smart ways to block all ip connections by default and add connection hostname/ip that is allowed to call in and make a call? |
| 17:45.24 | [TK]D-Fender | raidghost: man iptables <- |
| 17:46.58 | robl^ | or if you are using Ubuntu / Debian, "ufw" (which is a simpler front-end to iptables) |
| 17:48.52 | jmetro | raidghost: or dump all peers into default and only set registered peers to your inbound |
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| 17:55.24 | *** join/#asterisk Assid (~assid@unaffiliated/assid) |
| 17:55.25 | Assid | heya |
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| 17:56.07 | *** join/#asterisk clh (~clh@38.110.17.31) |
| 17:56.30 | clh | is there a doc bot on this channel? |
| 17:56.42 | Assid | so i need help with a decision i need to make.. im setting up a new location.. and i cant really use asterisk there.. however, i am looking for any sip based solution from any major company |
| 17:56.49 | Assid | like a polycom or something |
| 17:57.02 | Assid | any suggestions on a PBX based system ? |
| 17:57.07 | [TK]D-Fender | clh: Yes. What do you need to know? |
| 17:57.15 | WIMPy | You want SIP but no Asterisk? |
| 17:57.36 | clh | [TK]D-Fender: just wanted to show a friend what an IRC bot does |
| 17:57.53 | Assid | WIMPy: im open to asterisk as well.. but not something i make /setup.. something embedded like how legacy pbx systems work |
| 17:58.16 | [TK]D-Fender | infobot: areyouadog ? |
| 17:58.16 | infobot | Bark! Bark! |
| 17:58.26 | [TK]D-Fender | ~botsnack |
| 17:58.26 | infobot | :), [TK]D-Fender |
| 17:58.27 | WIMPy | And you think you can get that with SIP? |
| 17:58.33 | [TK]D-Fender | infooGood Boy ! |
| 17:58.37 | [TK]D-Fender | infobot: Good Boy ! |
| 17:58.37 | infobot | :), [TK]D-Fender |
| 17:58.47 | [TK]D-Fender | clh: Thre |
| 17:58.49 | [TK]D-Fender | there* |
| 17:58.58 | clh | thanks :) |
| 17:59.09 | Assid | WIMPy: personally i prefer asterisk.. but i ineed one of those pbx ina box type solutions |
| 17:59.10 | clh | infobot : tell me about sip |
| 17:59.27 | clh | infobot: chan_sip |
| 17:59.37 | [TK]D-Fender | clh: Don't expect too much... |
| 17:59.46 | clh | :) I see |
| 17:59.51 | clh | infobot: Good Boy! |
| 17:59.51 | infobot | :), clh |
| 17:59.56 | WIMPy | Assid: You know that Digium sells boxes? |
| 18:00.00 | [TK]D-Fender | clh: You wanted a sample from a bot... getting to ask what YOU want the WAY you want is another matter L( |
| 18:00.05 | *** join/#asterisk DarthExpeditor (~IceChat9@75.77.152.34.nw.nuvox.net) |
| 18:00.06 | [TK]D-Fender | WIMPy: That all use Asterisk |
| 18:01.05 | Assid | WIMPy: i couldnt find a vendor for dubai |
| 18:01.13 | Assid | or rather in |
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| 18:02.11 | [TK]D-Fender | Assid: So "not asterisk", and "vendor in dubai" ..... perhaps you should just check who resells there already... |
| 18:02.22 | [TK]D-Fender | Assid: Cisco can speak SIP.... and I'm sure they sell there. |
| 18:02.55 | Assid | the ones i found so far are ericsson-lg .. it does sip.. but charge per sip license/trunk |
| 18:02.57 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 18:02.58 | WIMPy | I'm sure so does SEN or Alcatel. |
| 18:03.19 | robl^ | Assid: Dubai / UAE is odd with VoIP. We have a ABu Dhabi office and we are not permitted to use VoIP outside of the office. In the building it is fine, but we can't use any sort of VoIP over the Internet. |
| 18:03.37 | WIMPy | They all do. So why do you want SIP? |
| 18:04.03 | Assid | WIMPy: softphone on cell for example . |
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| 18:04.38 | [TK]D-Fender | Assid: You keep piling on requirement after the fact. Stop making us run into them blind and jsut TELL US all of these rules it has to fit... |
| 18:04.42 | Assid | robl^: so what happens if you use it over the the internet |
| 18:05.24 | robl^ | Assid: our ISP blocks it. VoIP traffic will not flow in or out of our Abu Dhabi office |
| 18:05.32 | WIMPy | You find yourself in a stinking hole with bars? |
| 18:05.46 | Assid | robl^: and vpn ? |
| 18:06.36 | robl^ | Assid: VPN almost works. WE have issues with latency over VPN, as it would route to the USA. |
| 18:06.38 | Assid | [TK]D-Fender: ok.. i currently have a solution from ericsson .. im not going with it.. cause A> its a bit expensive B> sip license per extn/trunk C> i would like to use standard sip compliant devices instead of the more locked down proprietory devices which turn out to be expensive |
| 18:07.46 | [TK]D-Fender | Guess what... that's what almost all of the proprietary vendors do ... they make you PAY for everything. You are asking for the sort of things that aren't released in that commercial manner |
| 18:07.49 | WIMPy | With commercial PBXs the cost for proprietary clients which work is about the same as a VOIP licens with no guarantees whtsoever. |
| 18:07.56 | [TK]D-Fender | So what's the problem with it being Asterisk-based? |
| 18:08.56 | Assid | [TK]D-Fender: i wasnt able to find a vendor who supports it |
| 18:09.12 | Assid | robl^: your using asterisk based solution in abu dhabi ? |
| 18:09.46 | [TK]D-Fender | Oh, so not only does it have to be SOLD in Dubai, it also has to be SUPPORTED on-site there? Just go contact vendors there yourself then. You've pinned this to an obscene point. |
| 18:11.19 | talntid | wait, someone in dubai is complaining about something being a bit expensive? |
| 18:11.19 | robl^ | Assid: No. In Abu Dhabi, we have an Avaya IP Office solution. We also deploy Nortel Multimedia PC Clients (which use SIP and connect to Nortel MCS 5200 in the US). However neither will work over the internet for us in AD. |
| 18:11.23 | talntid | how much was it? |
| 18:14.08 | Assid | [TK]D-Fender: preferably good to have support.. so i dont have overheads.. generally the bigger brands are sold there.. |
| 18:14.55 | [TK]D-Fender | Assid: Just go contact vendors then. This shouldn't be a guessing game since you already know who isn't sold there. |
| 18:15.05 | malcolmd | Assid: no vendor in dubai? here's FVC (http://www.digium.com/en/partners/distributors/fvc-inc), and here's DVCOM (http://www.digium.com/en/partners/distributors/dvcom-technology). i don't work on the sales side but one of them probably has an integrator that works in the same region that does on-site support |
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| 18:16.32 | Assid | hm interesting.. last time i tried i didnt get very far.. |
| 18:16.45 | talntid | clearly just a lack of research. Doesn't want to research, but doesn't want to pay the prices of the knowledgable vendors |
| 18:16.50 | [TK]D-Fender | When was that? |
| 18:16.58 | Assid | last week.. |
| 18:17.40 | Assid | wait.. i wasnt really looking for digium/asterisk as such..more along the lines of polycom etc |
| 18:19.10 | talntid | Assid, what is your role at this company? |
| 18:19.26 | Assid | let me see what FVC offers |
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| 18:45.14 | robert_ | so AST keeps trying to connect to LAN IP's |
| 18:46.32 | robert_ | "ERROR[17412] tcptls.c:446 ast_tcptls_client_start: Unable to connect SIP socket to 10.0.25.151:5855: Connection timed out" |
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| 18:48.56 | robert_ | igcewieling: any ideas? |
| 18:49.54 | robert_ | [TK]D-Fender: sup |
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| 18:53.43 | [TK]D-Fender | robert_: Idea : look at your configs and prove * is even listening for it |
| 18:54.21 | robert_ | also |
| 18:55.32 | robert_ | I get mysql timeouts when doing 'sip show peer 2000 load' :/ |
| 18:55.47 | robert_ | I'm thinking I might switch to sqlite, lol |
| 18:58.06 | robert_ | [TK]D-Fender: where am I looking? |
| 18:59.47 | [TK]D-Fender | robert_: Your sip configs |
| 18:59.59 | robert_ | where in my sip config, though? |
| 19:00.40 | [TK]D-Fender | robert_: how about the part where you even told SIP is permitted to use TCP? This is your server. How do you not know this? |
| 19:01.18 | [TK]D-Fender | robert_: And you chould be looking at your calls closer as well. |
| 19:01.51 | robert_ | it's been a while since I touched our AST config, lol |
| 19:02.03 | robert_ | SIP is using TLS |
| 19:03.35 | [TK]D-Fender | robert_: You pasted a single line from debug and aren't showing a full communication. We can only guess what the rest shows. |
| 19:05.05 | robert_ | hm |
| 19:11.23 | robert_ | [TK]D-Fender: hold and I'll paste sip.conf |
| 19:14.49 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.211) |
| 19:16.24 | robert_ | [TK]D-Fender: http://pastebin.com/B4fMHFcT |
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| 19:20.56 | robert_ | (please ignore the date; I simply reuse posts I don't need anymore.) |
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| 19:53.50 | robert_ | [TK]D-Fender: hi. any ideas? |
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| 20:05.33 | [TK]D-Fender | robert_: Why aren't you looking at the complete communication? |
| 20:05.41 | [TK]D-Fender | robert_: You said it wasn't working as expected |
| 20:05.53 | [TK]D-Fender | robert_: You didn't show us a whole communication to comment on. |
| 20:06.03 | [TK]D-Fender | robert_: You don't seem to be actually looking at the problem. |
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| 20:54.41 | DBordello | Does anybody have any recommendations for a windows softphone? |
| 20:55.01 | igcewieling | All softphones suck |
| 20:55.11 | carrar | xlite |
| 20:55.16 | DBordello | That may be, but I am still waiting for my hardphone to arrive |
| 20:55.46 | DBordello | carrar, thanks for the recommendation |
| 21:03.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:14.45 | _Corey_ | DBordello: X-Lite has become very ad-oriented as of late... you may want to look at something like Jitsi or perhaps Zoiper if you want one that resembles a phone. |
| 21:15.11 | DBordello | MicroSIP looks pretty light weight |
| 21:16.25 | _Corey_ | Don't know that one, but it sounds like you're heading in the right direction |
| 21:16.40 | jmetro | I like 3cx phone |
| 21:16.51 | jmetro | the free one |
| 21:17.25 | DBordello | I'll check it out, thanks |
| 21:18.00 | DBordello | 3CX looks nice |
| 21:18.11 | *** join/#asterisk przerull (~philip@50.56.205.232) |
| 21:18.35 | przerull | hello, so why isn't HANGUPCAUSE being set after a dial command in asterisk 11? |
| 21:19.51 | [TK]D-Fender | przerull: show us |
| 21:20.56 | przerull | D-Fender: would you like me to paste the dumpchan and a dialplan snippit? |
| 21:21.15 | *** join/#asterisk clh (~clh@107-202-133-88.lightspeed.tukrga.sbcglobal.net) |
| 21:21.40 | przerull | granted, it might have something to do with the fact that I'm trying to get it from inside a hangup handler |
| 21:22.12 | igcewieling | pastebin the CLI output of this running: Noop(HANGUPCAUSE is '${HANGUPCAUSE}') |
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| 21:26.41 | przerull | hmmmm it was defiately there, the hangupcause for my inbound leg didn't match that for my outbound leg but that's not too big of an issue |
| 21:27.23 | przerull | i must have a typo somewhere (not enough sleep). Thanks |
| 21:32.21 | gorkish | what would be the recommended approach to building a simple dial-by-voice directory app? Full res_speech + unimrcp + lumenvox subscription licenses or is there something that would work as well that is simpler or free just for this limited use case |
| 21:32.56 | apb1963 | DBordello: phonerlite |
| 21:33.15 | DBordello | apb1963, thanks :) |
| 21:33.42 | apb1963 | let us know what you choose in the end |
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| 21:37.17 | DBordello | apb1963, will do. |
| 21:38.55 | apb1963 | I like it because it seems to be full featured, including conferencing and recording. |
| 21:39.27 | robl^ | you can always go with a paid version of Bria softphone |
| 21:39.37 | apb1963 | oh and it's free :) |
| 21:40.15 | apb1963 | supports many codecs, echo cancellation, sampling.... |
| 21:41.41 | apb1963 | and a few things I have no idea what they do :) |
| 21:42.42 | apb1963 | Recording was a bit of a bear to find... it's only available when a call is in progress... you have to right click on the active call. Drove me crazy until I nailed it down. |
| 21:43.57 | apb1963 | I have however returned to my former semi-sane self |
| 21:46.02 | apb1963 | resumes sharpening an onion |
| 21:48.40 | *** join/#asterisk Kraln (~kraln@69.169.90.240) |
| 22:07.40 | DBordello | What should I use for the domain when configuring a phone? |
| 22:08.05 | navaismo | domain of asterisk or ip |
| 22:08.41 | DBordello | great, thanks :) |
| 22:09.56 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
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| 22:25.32 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
| 22:26.17 | navaismo | I wish all irc are very helpful as this :'( even with third party addons |
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| 22:45.27 | DBordello | I am playing with my first Asterisk install, using FreePBX on a Raspberry Pi |
| 22:45.36 | DBordello | It is quickly becoming apparent how far in over my head I am :) |
| 22:47.41 | navaismo | uh dont tell Freepbx here they hate that and you will receive the ~freepbx |
| 22:49.35 | DBordello | That is fair enough. |
| 22:49.51 | DBordello | Right now my biggest issue is not understanding the fundamentals, not the configurations |
| 22:53.29 | navaismo | fundamentals? |
| 22:54.19 | DBordello | trunks/outgoing routes/incoming routes |
| 22:56.12 | navaismo | most items has their help with the icon ? |
| 22:56.36 | DBordello | Yup, doing some reading now. |
| 23:04.18 | jmetro | anyone feel like telling me what to fix in my queue |
| 23:04.39 | jmetro | everything works except the queue announcement announces at the beginning , when i change it , it doesnt announce at all. |
| 23:06.29 | *** join/#asterisk italorossi (~italoross@187.61.168.117) |
| 23:10.19 | jayk | <PROTECTED> |
| 23:10.21 | jayk | oops |
| 23:10.38 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 23:11.37 | WIMPy | Midnightdan |
| 23:12.05 | jmetro | noone wants to fix my stupid queue announcements? |
| 23:17.02 | igcewieling | jmetro: queues are beyond even my nearly mystic skills. |
| 23:18.07 | jmetro | arent they the most fucking broken thing? i mean cmon =\ |
| 23:18.36 | jmetro | i dont...what... |
| 23:18.53 | jmetro | i just set all of my announcements to the same file.. and now it does it correctly. |
| 23:19.29 | jmetro | i should be hearing this thing played 5 times in a row. |
| 23:25.15 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 23:25.28 | apb1963 | jmetro: I know absolutely nothing about queues.... but if one file plays and another doesn't... I'd look at file permissions. <shrug> |
| 23:27.04 | danfromuk | WIMPy: were you talking to me? |
| 23:28.13 | WIMPy | danfromuk: Yes ;-) |
| 23:28.46 | danfromuk | No rest for the hard working :-) |
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| 23:30.31 | jmetro | apb1963: nah the files are playing just fine, i rename them to occupy the slot i want them to be in [midqueue rather than intro] and they dont play, i move them again and they play but only at the beginning. |
| 23:30.33 | jmetro | my solution was this. |
| 23:30.40 | jmetro | queue-thankyou=queue-periodic-announce |
| 23:30.40 | jmetro | queue-callswaiting=queue-periodic-announce |
| 23:30.40 | jmetro | queue-thereare=queue-periodic-announce |
| 23:30.40 | jmetro | queue-youarenext=queue-periodic.announce |
| 23:30.40 | jmetro | periodic-announce=queue-periodic-announce |
| 23:30.54 | jmetro | queue-periodic-announce is my real announcement overwriting the file. |
| 23:31.11 | jmetro | plays it every min-announce-freq |
| 23:31.22 | jmetro | once. not 5 times like it should be. |
| 23:31.50 | jmetro | needless to say, the people who wrote app_queue were the same folks who wrote the cisco web ui's. Very angry people living in new delhi |
| 23:32.27 | WIMPy | jmetro: Don't throw yourself in the bin. |
| 23:33.08 | jmetro | WIMPy: ? |
| 23:33.20 | WIMPy | ~pb |
| 23:33.20 | 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. |
| 23:33.33 | jmetro | oh yeah, i realized that after i sent the multi-liner |
| 23:33.39 | WIMPy | Throw your data on a bin instead. |
| 23:33.42 | jmetro | it all looks like one post in pidgin |
| 23:34.25 | WIMPy | That triggered an auto-ignore for me. |
| 23:36.32 | jmetro | http://pastebin.com/LPK8FN0c if you want a comparison |
| 23:36.49 | WIMPy | I have never used queues. |
| 23:37.14 | jmetro | WIMPy: you must have at least 30 less gray hairs than me. |
| 23:37.18 | WIMPy | I just read in to them once and was rather disappointed at what I read. |
| 23:37.30 | WIMPy | Because they fell out :-) |
| 23:37.36 | *** join/#asterisk infernix (~nix@unaffiliated/infernix) |
| 23:38.16 | jmetro | Now i have to go clean up my own code and wonder if theres a better way that i can handle two code flows that are the exact same except one gets call announcements. |
| 23:38.26 | apb1963 | omg.... I just realized... Wimpy was the hamburger eating guy on Popeye. lol |
| 23:38.54 | WIMPy | Jepp. :-) |
| 23:39.09 | apb1963 | Now we know what you look like. lol |
| 23:39.30 | WIMPy | Yes, I look like hin more and more. |
| 23:40.02 | jmetro | I'll gladly pay you Tuesday for a findme/followme code today! |
| 23:40.49 | WIMPy | You don't like the app? |
| 23:41.15 | jmetro | theres an app for asterisk? |
| 23:41.44 | WIMPy | core show application followme |
| 23:41.50 | jmetro | oh jesus |
| 23:42.15 | jmetro | yeah we dont have that installed but it looks nice |
| 23:42.27 | WIMPy | Surprise: There are a few things you don;t have to build yourself. |
| 23:43.25 | jmetro | Mines based on statuses from isymphony though |
| 23:43.29 | jmetro | so that would get sticky. |
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| 23:57.46 | phix | 1.8 still supported? |