| 00:14.53 | *** join/#asterisk splnet (~splnet@128.239.28.2) |
| 00:18.58 | NightMonkey | Hi all. I'm looking at testing/playing around with Freenum, for which I just got a number assigned. However, up to now my asterisk has been safely tucked behind NAT. |
| 00:20.35 | NightMonkey | Information seems to be scattered on properly securing Asterisk (or VoIP services) on the Internet. Is it generally better to put some sort of SIP proxy on the Internet, that proxies for Asterisk, and how do you mitigate simple DoS attacks? |
| 00:24.25 | *** part/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 00:30.21 | [TK]D-Fender | NightMonkey: SIP attacks would be via something like fail2ban. Everything else is basic firewalling |
| 00:32.31 | NightMonkey | [TK]D-Fender: Thanks. |
| 00:42.45 | splnet | Does anyone know of any free SIP providers that are providing Skype peering? |
| 00:46.35 | pabelanger-lap | splnet: just get the SFA channel driver and set it up yourself |
| 00:48.31 | Docteh | "skype peering"? |
| 00:48.39 | splnet | pabelanger-lap: not familiar with that, but I'm assuming that requries a skype client on the same box as the asterisk server does it not? |
| 00:49.00 | splnet | Skype has had inbound SIP access for a while now |
| 00:49.01 | pabelanger-lap | ~skypeforasterisk |
| 00:49.02 | infobot | extra, extra, read all about it, skypeforasterisk is 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 |
| 00:49.09 | pabelanger-lap | splnet: ^^ |
| 00:49.38 | splnet | only available for select partners though |
| 00:50.01 | splnet | or at least it was several years ago |
| 00:50.01 | pabelanger-lap | splnet: what partners? |
| 00:50.18 | pabelanger-lap | SFA is an asterisk channel driver |
| 00:50.46 | splnet | pabelanger-lap: how many simultaneous channels does it support? |
| 00:51.06 | pabelanger-lap | licenses is 1 to 1 ratio |
| 00:51.16 | pabelanger-lap | So, 10 licenses is 10 channels |
| 00:51.20 | Docteh | check out the webpage on it |
| 00:51.58 | splnet | cool! looks interesting |
| 00:54.37 | splnet | What I was talking was different. Sip providers could apply to skype to get a special SKYPE only DID that would allow the skype network to call IN to your system . |
| 00:58.30 | Docteh | so skype user punches in a number or a username? |
| 00:58.36 | ChannelZ | maybe I'm missing your subtlety but that's what SFA can do. Bridge Skype in and out of Asterisk. |
| 00:59.06 | splnet | Docteh: right a special number |
| 01:00.04 | splnet | ChannelZ: the difference is, with the method I'm talking about, all you need is SIP. SKype does the translation |
| 01:00.44 | splnet | Anyhow, this was several years ago.. they may not do that anymore |
| 01:00.45 | Docteh | that might end up being more expensive than skype for sip, which they're offering in beta |
| 01:00.50 | ChannelZ | Skype has their own SKype for SIP product |
| 01:01.13 | Docteh | well if you present them with a pile of money i'm sure they'll do whatever you want ;) |
| 01:01.50 | splnet | A startup company I was working for in 2005 got one of these numbers. We were trying to rollout a skype for sip like product |
| 01:01.53 | ChannelZ | With Skype For SIP you pay some monthly fee plus whatever minutes. With SFA you pay once for a channel and any minutes |
| 01:02.06 | ChannelZ | So it's not like you're saving a ton of money buy not doing it yourself |
| 01:02.20 | ChannelZ | s/buy/by |
| 01:03.29 | splnet | ChannelZ: so remote users are forced to hairpin audio through SFA right? |
| 01:06.16 | Docteh | the asterisk server would be running skype for asterisk |
| 01:06.53 | ChannelZ | yeah |
| 01:08.48 | splnet | Right What I meant was: SIP asterisk user a talking to skype user b would be forced to route RTP through the asterisk server, I'm assuming |
| 01:08.56 | *** join/#asterisk jblack (~jblack@71.181.244.180) |
| 01:09.26 | ChannelZ | yes |
| 01:24.59 | *** join/#asterisk Mango (~iMango@d154-20-89-230.bchsia.telus.net) |
| 01:25.57 | NightMonkey | Any tips on methods to have asterisk ignore "silent" voicemails? Callers who hang up within seconds of voicemail answering leave 5-7 second recordings of silence or dial-tone, and I'd like to reduce those. |
| 01:26.19 | NightMonkey | Even better would be to still generate an e-mail, but leave off the attached file. |
| 01:26.59 | NightMonkey | (This is when using the SPA3102 between asterisk and POTS) |
| 01:28.34 | Mango | Is the XML Microbrowser supported on the Polycom IP 320/330? I found one manual that said it was and one that said it wasn't. |
| 01:29.10 | NightMonkey | Mango: I have it "working" (but it's a bear) |
| 01:30.26 | Mango | oh? |
| 01:30.51 | NightMonkey | Mango: Just a simple weather page, but spacing is quite tight. |
| 01:31.05 | Mango | ok, so not much room on the screen? |
| 01:31.22 | NightMonkey | Mango: Right. And the supported syntax is rather anemic. |
| 01:31.50 | NightMonkey | Mango: But the feature is there. At least with relatively recent firmware. |
| 01:32.28 | Mango | Firmware is free, right? |
| 01:33.44 | NightMonkey | In cost, if not in spirit. |
| 01:33.51 | Mango | Heh. |
| 01:34.08 | Mango | There's a bunch on eBay right now. I'm thinking of picking one up. $60; not bad. |
| 01:35.17 | Docteh | I've been emailing myself spectrograms of messages |
| 01:35.24 | NightMonkey | Mango: Don't expect to have much fun with the screen, but as a phone, it's been very solid over several years. |
| 01:35.48 | Mango | as for your question, you may want to experiment with silencethreshold and maxsilence |
| 01:36.22 | NightMonkey | Mango: Thanks. |
| 01:36.24 | Docteh | do either of those cause the recording to get thrown out? |
| 01:36.36 | Docteh | or does it just make asterisk give up quicker |
| 01:37.12 | Mango | My wife has a very soft voice. |
| 01:37.26 | Mango | If improperly configured, the recording will get thrown out. |
| 01:40.26 | Mango | It rather annoys her. |
| 01:40.28 | Mango | ;) |
| 01:41.58 | NightMonkey | Hrm. Those get me part way there, but it doesn't appear to cause the message to be abandoned before adding it to a mailbox. |
| 01:42.20 | Mango | :-/ |
| 01:43.12 | NightMonkey | Mango: No worries. Not a big deal, but just annoying. |
| 01:44.13 | NightMonkey | Mango: Have you worked with Polycom phones before? |
| 01:44.31 | Mango | No, I have not. |
| 01:44.42 | Mango | Any gotchas? |
| 01:45.23 | NightMonkey | Mango: Firmware updates, IIRC, are only via tftp/bootp and the configs are held in XML files. Not all config options are available through the phone's web server, or menus. |
| 01:46.02 | Mango | That's minorly annoying, but I can live with it. |
| 01:46.15 | NightMonkey | Mango: The XML files, at least for me, are hell to work with. Giant long many-screen'd lines make vi kinda unhappy. |
| 01:46.31 | Mango | I'm used to it...I write webapps. |
| 01:46.48 | Mango | No less hell, but at least it's the same hell :) |
| 01:47.15 | NightMonkey | Mango: Then you're set. |
| 01:47.19 | *** join/#asterisk DarkRift (~dark@modemcable219.40-56-74.mc.videotron.ca) |
| 01:47.34 | Mango | What I'm really after is the multiple registrations |
| 01:47.40 | Mango | My SPA921 only supports one registration. |
| 01:47.48 | NightMonkey | Mango: For multiple lines? |
| 01:47.52 | Mango | Server down? Oh! That's too bad! |
| 01:48.09 | Mango | For redundancy |
| 01:48.22 | NightMonkey | Mango: Oh. I haven't tried that, but I *think* that's doable. |
| 01:48.31 | Mango | Yeah, I've been reading the admin guide and it seems to be. |
| 01:48.53 | Mango | With the SPA921, you can configure priority based on a SRV record, but it only registers to the backup after the primary has been confirmed down. |
| 01:49.09 | Mango | With the Polycom, you register to both at once, so there is no downtime. |
| 01:49.42 | Docteh | how long does it take the phone to realize the server is down? |
| 01:50.37 | Mango | It depends on the registration interval. |
| 01:50.49 | NightMonkey | Mango: Gosh, that's a dim way of offering "backup" functionality. |
| 01:50.57 | Mango | aye :) |
| 01:51.10 | Mango | When I bought this phone I didn't know anything about VoIP. So I didn't know to investigate that. |
| 01:51.16 | Mango | I just went, "CTU Ringtone! Cool!" |
| 01:52.37 | NightMonkey | How can Cisco be so smart in some areas and so dim in others? Too much left brain, not enough right? |
| 01:53.33 | Mango | I think this might be a rebranded Sipura |
| 01:53.49 | Mango | I guess they wanted people to buy the higher-end phone that did have that feature. |
| 01:54.30 | Mango | No problem if you have your SIP servers on the same LAN as the phone; then there's no need for registration. |
| 01:56.55 | Docteh | didn't linksys buy sipura? |
| 01:57.37 | Docteh | my SPA3102 is a Linksys by Cisco version |
| 01:58.03 | Mango | Yeah |
| 01:59.03 | Mango | Technically, Cisco bought Sipura and gave it to Linksys as a present. |
| 01:59.07 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 02:21.44 | *** join/#asterisk coppice (~chatzilla@210.17.254.185) |
| 02:28.13 | ChannelZ | If that's true Linksys gave it back because all Linksys is doing is home routers and things now. |
| 02:29.26 | Mango | I don't recall any major developments in Linksys products |
| 02:29.34 | Mango | *Linksys VoIP products |
| 02:29.49 | Mango | most of them stayed the same, didn't they? they just got rebuilt to look like Cisco |
| 02:30.07 | Mango | My phone's admin page still says "Sipura SPA Configuration" |
| 02:31.52 | Crymsonite | I was able to make a SIP call to my Server B, and ring my sip soft phone. But the only way I could get it to work, was to add an extension, and Dial the SiP device locally. DIAL SIP/ServerB/SIPPhone1, it treats the SIPPhone1 as a extension to look up on Server B. Is there anyway, I can just tell server B to dial the device named SIPPhone1, not go to the extension SIPPhone1? Or is there a clever easy way to acheive what I want? |
| 02:32.39 | ChannelZ | I dunno. I have a bunch of SPA922s that are all branded Linksys but all that's on Linksys' site anymore is this 'Valet' thing, although the pictures have Cisco on them. No VoIP stuff appears on linksys.com last I looked |
| 02:32.40 | Crymsonite | Sorry thats wrong, I did Dial SIPPhone1@ServerB |
| 02:33.08 | ChannelZ | Cisco lists all the SPA phones as 'Cisco SPA9xx' yet all their pictures still say Linksys. So who the hell knows. Identity crisis |
| 02:34.36 | ChannelZ | Crymsonite: Server A has an extension 111 which does a Dial(SIP/ServerB/222) and ServerB's extension 222 does a Dial(SIP/softphone).... |
| 02:35.38 | ChannelZ | assuming 'softphone' is registered to ServerB that's about how it aught to work |
| 02:35.47 | Crymsonite | yes, exactly |
| 02:36.12 | Crymsonite | I was not looking forward to writing to the extensions.conf and sip.conf all 200 sip phones that I would use in production. |
| 02:37.29 | ChannelZ | if you classify them, you don't have to. IE if ServerB has 200 phones and all the extensions are similar (like 1xxx), you can just use a pattern. |
| 02:37.29 | Crymsonite | Wonder if theres a way I can pass a variable to ServerB at the sametime, to help out. |
| 02:38.54 | Crymsonite | Like naming them 1SIPPhone1, 1SIPPhone2, 1SipPhone3, use a pattern match, and strip the first one and do the dial? |
| 02:39.00 | Crymsonite | That just might be it. :D |
| 02:39.18 | ChannelZ | well no |
| 02:39.32 | ChannelZ | extensions are generally just numbers. |
| 02:39.37 | ChannelZ | Devices have names. |
| 02:40.15 | ChannelZ | As in exten => 100,1,Dial(SIP/JoesPhone) |
| 02:40.33 | ChannelZ | Dial 100, it calls JoesPhone |
| 02:41.09 | Crymsonite | Gotchya. |
| 02:41.28 | Crymsonite | ServerB wouldn't retain any of the variables used by the call from ServerA would it? |
| 02:41.37 | ChannelZ | If you want to not have to write a dialplan for each person, the easiest way probably is to make all your SIP device names the same as their 'extension' |
| 02:42.52 | Crymsonite | Yeah, that makes perfect sense. Just making sure theres no shortcut way of doing what I want before I write it off on my want list. |
| 02:42.58 | ChannelZ | That way you can do exten => _1XXX,1,Dial(SIP/${EXTEN}) for instance.. someone dials 1234, asterisk calls SIP/1234 |
| 02:44.44 | Docteh | If I did that I'd prefix the sip usernames with something |
| 02:45.02 | ChannelZ | do whatever you want |
| 02:45.17 | ChannelZ | With Asterisk, You Can Do That(tm) |
| 02:51.07 | Crymsonite | Ah I just got it. Thank god for FastAGI. :D |
| 02:52.15 | *** join/#asterisk mrchrisadams (~Adium@121.127.221.124) |
| 02:55.53 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 02:58.15 | penguin_ice | hey where can i find a free-text-to-speech for my asterisk box to read out the news |
| 02:58.19 | penguin_ice | ??? |
| 02:58.28 | penguin_ice | not flite or swift they sound like shit |
| 02:59.22 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 03:01.18 | Docteh | get a mac and use the Alex voice? |
| 03:02.27 | penguin_ice | do i have to have a mac? |
| 03:03.03 | penguin_ice | cant i just get the alex voice into my linux machine? |
| 03:03.09 | pabelanger-lap | penguin_ice: sphinx? |
| 03:03.20 | Docteh | well theres a tts engine in recent OSX, I tried looking around but nobody seems to have tried hacking it onto linux |
| 03:03.53 | penguin_ice | sphinx? |
| 03:08.09 | ChannelZ | What, you no likes Festival? |
| 03:09.12 | Docteh | penguin_ice: I had better results using the regular swift engines instead of the special 8khz ones |
| 03:09.23 | pabelanger-lap | penguin_ice: nevermind, sphinx is ASR |
| 03:13.55 | Docteh | I wonder how hard it'd be to run darwin binaries on linux |
| 03:30.43 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 03:39.17 | *** join/#asterisk DrCron (rszasz@saxonco.com) |
| 04:04.01 | coppice | penguin_ice: swift is as good as anything out there, but many of the Cepstral voices are poor. Try swift with the callie voice, and its pretty reasonable. I don't know anything better (except in demos - demos usually sound wonderful) |
| 04:05.38 | *** join/#asterisk DogBoy (~john@unaffiliated/dogboy) |
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| 04:46.44 | Docteh | the apple one reminds me of Doctor Manhatten from the Watchmen movie |
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| 05:53.39 | ChannelZ | Is that naked glowing guy? |
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| 06:30.14 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 06:31.13 | raden | heya Naikrovek |
| 06:31.17 | raden | morning Katty |
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| 06:51.18 | redax | good morning, |
| 06:51.36 | raden | morning |
| 06:52.49 | redax | don't you know what makes my 1.6.2.9 asterisk losts sip phones... I had to downgrade to 1.6.0.x |
| 06:53.06 | redax | after 1-2 days it get loose more and more sip client |
| 06:53.29 | redax | after asterisk restart everything's fine again for a few hours/days |
| 06:57.24 | raden | sip debug ? |
| 06:57.56 | redax | not yet, my collegue just reported the problem |
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| 08:07.32 | Godfather_ | o/ |
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| 08:24.26 | ndemir | hello. what is the differences between softswitch and IP PBX? (freeswitch vs asterisk)? |
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| 12:12.15 | dauergast | hi is it possible to playback a sound file to the channel, after I hang up? |
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| 12:13.56 | [sr] | howdy |
| 12:15.37 | dauergast | when i hangup before the person on the other side does, can I use the hangup extension? |
| 12:16.41 | florz | sure you can - "hangup" is just another word for "noop" |
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| 13:08.08 | dauergast | so hangup extension is triggered when caller, OR callee hang up? |
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| 14:35.44 | N|Xgurru | hi everyone |
| 14:35.54 | N|Xgurru | need help |
| 14:36.11 | N|Xgurru | I am having one issue in Asterisk voice recording with record() function |
| 14:38.04 | N|Xgurru | when someone records his voice through sip that last few seconds of voice gets recorded fast |
| 14:38.15 | N|Xgurru | means last few seconds of voice is fast |
| 14:38.46 | N|Xgurru | anyone has any such issues in Asterisk 1.4.25.1 ? |
| 14:40.31 | N|Xgurru | anyone there pls ??????? |
| 14:57.43 | troy42 | nobody here but us bots |
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| 15:09.27 | N|Xgurru | need help in asterisk record fast issue? |
| 15:09.34 | N|Xgurru | kindly someone reply |
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| 16:36.53 | *** join/#asterisk csd-199 (~asdf@189.237.37.68) |
| 16:37.56 | csd-199 | hi to all. I just installed asterisk 1.6 in CentOS using YUM... but it didn't install the ilbc codec, How can I install the ilbc codec? |
| 16:39.41 | [TK]D-Fender | csd-199: has to be compiled seperately and has prerequisites |
| 16:40.18 | csd-199 | uff that means I have to do all manualy, right? |
| 16:44.59 | *** join/#asterisk pabelanger (~pabelange@yoda.kanatek.com) |
| 16:49.53 | [TK]D-Fender | csd-199: Cruise the repo to see if its there somewhere |
| 16:50.36 | carrar | Manually is the best way |
| 16:51.42 | carrar | Otherwise it's like going to fast food, you really never know whats inside your food |
| 16:52.17 | rajiv | can someone recommend an origination provider in the US for residential+asterisk use? ideally someone with unlimited incoming minutes. |
| 16:52.51 | carrar | ~itsp |
| 16:52.52 | 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. |
| 16:53.12 | rajiv | ~itsplist-us |
| 16:53.13 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 16:53.22 | rajiv | carrar: thx |
| 17:13.28 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
| 17:19.03 | rajiv | how about a company in the boston, ma, usa area who can setup and manage an asterisk pbx for a small business ? |
| 17:20.36 | [TK]D-Fender | rajiv: Wouldn't that be YOU? |
| 17:21.35 | rajiv | [TK]D-Fender: funny. i have one client i'm managing but need to step away from that. they have an asterisk box i built for them but its time to upgrade |
| 17:30.34 | [TK]D-Fender | rajiv: Foster an admin :) |
| 17:35.35 | Godfather_ | is possibly to know who hang up the channel ? (caller or callee) |
| 17:41.00 | *** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus) |
| 17:47.01 | [TK]D-Fender | Godfather_: g <----------- |
| 17:49.16 | *** join/#asterisk lost_soul (shackett@devio.us) |
| 17:49.41 | *** join/#asterisk path (path@gateway/shell/bshellz.net/x-tftmraijhnzlmloa) |
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| 19:04.58 | *** join/#asterisk Godfather_ (~Godfather@193.153.129.150) |
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| 19:06.42 | Docteh | ~itsplist-ca |
| 19:06.43 | infobot | Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca , http://www.voip.ms |
| 19:11.23 | *** join/#asterisk shtoom (~shtoom@115.118.76.242) |
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| 19:35.01 | *** join/#asterisk lbt (~david@Maemo/community/contributor/lbt) |
| 19:37.51 | lbt | Hi... just wanted to mention that I've made a clone at gitorious for the UK-CLID patches that have been floating around for a few years : http://www.voip-info.org/wiki/index.php?page_id=5781&tk=22ea721a5916a04da74f&comments_page=1 ... just an FYI and maybe for google for UK caller ID |
| 19:38.06 | lbt | http://gitorious.org/~lbt/asterisk-tools/lbts-asterisk-with-uk-clid |
| 19:38.12 | lbt | http://gitorious.org/~lbt/asterisk-tools/lbts-dahdi-linux-with-uk-clid |
| 19:38.50 | lbt | cheers... feel free to irc pm me |
| 19:38.55 | *** part/#asterisk lbt (~david@Maemo/community/contributor/lbt) |
| 19:43.20 | *** join/#asterisk p3nguin (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 19:50.23 | *** join/#asterisk seanjohn (~seanjohn@gateways.sheltoncomputers.com) |
| 19:53.25 | seanjohn | qualify=yes or qualify=100 was what I THOUGHT was the time, in ms, it took to connect the call and then go to the next trunk in context if it couldn't connect, for instance 10000, in 10 seconds. I have now realized this is actually the maximum delay allowed for each trunk before marking it unusable. WHAT config parameter actually does the time to connect a call, in sip.conf. My trunks have always been in sequential order of execution f |
| 19:53.25 | seanjohn | least latent to the most and NONE are over 80ms. |
| 19:54.48 | seanjohn | I have already set the default of qualify to 100ms and all trunks now have qualify=yes instead of a time definition |
| 19:56.21 | [TK]D-Fender | seanjohn: Aualify is not a valid means of judging anything more than if the host is responding. |
| 19:56.47 | [TK]D-Fender | seanjohn: It is NOT viable for calculating latency in generla as OPTIOSN packets my get lower prioritization, etc. |
| 19:56.50 | seanjohn | yeah, i figured that out. Is there a parameter for sip.conf to limit how long it takes to connect a call? |
| 19:57.00 | [TK]D-Fender | deanjPolycom's look like 80ms on a local LAN |
| 19:57.12 | [TK]D-Fender | [15:56]<seanjohn>yeah, i figured that out. Is there a parameter for sip.conf to limit how long it takes to connect a call? <- no |
| 19:57.19 | seanjohn | I use linksys ATA's and they get 20ms on local lan |
| 19:57.37 | seanjohn | I think i'll set qualify to 200 |
| 19:57.40 | [TK]D-Fender | seanjohn: It also wouldn't be a PARAMETER, it'd just be a "statistic", and one dependent on your basis of valuation |
| 19:57.59 | [TK]D-Fender | seanjohn: leave Qualify at 2000+ is you know what's good for you |
| 19:58.18 | [TK]D-Fender | seanjohn: That doesn't set the frequency so it won't hurt you to save some breathing room |
| 19:58.33 | seanjohn | I know I was just hoping to have faster failover |
| 19:59.04 | [TK]D-Fender | seanjohn: If the host is considered "up" then the call will take some time to fail.... |
| 19:59.12 | [TK]D-Fender | seanjohn: Firmly in TFB territory |
| 20:00.21 | seanjohn | my trunks are reporting their ACTUAL time in ms through "sip show peers" but the ATA's are showing fake |
| 20:00.37 | seanjohn | 20ms to travel through 100 feet of cable? |
| 20:00.45 | seanjohn | lol |
| 20:01.29 | [TK]D-Fender | seanjohn: as I said this isn't true latency. So take that idea and BURN IT |
| 20:01.33 | WIMPy | seanjohn: 100ft of cable plus trhe software of the ata. |
| 20:01.36 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 20:02.37 | seanjohn | as far as the prioritization, I have tos_sip=cs3 and tos_audio=ef WHILE, at the same time, I have the firewall marking all packets originating from or going to ports 5060:5064 and 10000:20000 with 10000:20000 my RTP port settings |
| 20:03.10 | seanjohn | call quality is great |
| 20:03.13 | [TK]D-Fender | seanAll meaningless |
| 20:03.51 | seanjohn | uhhh TC (traffic control) shows the upstream prioritized |
| 20:04.37 | [TK]D-Fender | .... |
| 20:04.49 | seanjohn | class htb 1:30 parent 1:1 leaf 30: prio 3 rate 52000bit ceil 600000bit burst 6Kb cburst 1599b |
| 20:04.49 | seanjohn | <PROTECTED> |
| 20:04.56 | [TK]D-Fender | seanDevices can choose to RESPOND to OPTIONS with a lower priority <------ |
| 20:05.05 | seanjohn | that's the class asterisk falls into |
| 20:05.35 | [TK]D-Fender | seanjohn: The path may be incredible but the device says "OPTIONS? Fuck it, I don't care, you can WAIT and will report in SLOWER" |
| 20:05.57 | seanjohn | you're talking about the devices that are remote? |
| 20:06.12 | [TK]D-Fender | seanjohn: failover isn't going to trigger any faster |
| 20:06.18 | seanjohn | the media converter honors QoS |
| 20:06.32 | [TK]D-Fender | seanjohn: No, I'm talking about EVERY FUCKING DEVICE regardless of location <- |
| 20:06.42 | [TK]D-Fender | seanYou aren't getting it. |
| 20:07.03 | [TK]D-Fender | seanjohn: APPLICATION LEVEL LATENCY <---- Has JACK-SHIT to do with youre REAL NETWORK PERFORMANCE |
| 20:07.24 | [TK]D-Fender | seanjohn: Stop compareing apples an ORANGUTANS |
| 20:07.35 | *** join/#asterisk Obeliks (obeliks@gentoo/contributor/Obeliks) |
| 20:08.03 | seanjohn | I get what you're saying but, remotely testing 1000 miles away, the setup is fine with me and has no delay. Thanks for helping me with qualify. I set it to 2000 |
| 20:08.16 | [TK]D-Fender | seanjohn: Yes the packet arrives fast and I shove it on the BOTTOM of the to-do list. |
| 20:08.35 | seanjohn | I know I have no control of downsteam |
| 20:08.39 | seanjohn | downstream |
| 20:08.59 | [TK]D-Fender | seanjohn: I suggest a higher qualify because in the case of a freak slow response it will consider the peer down and burn a call attempt it shouldn't should one be requested. |
| 20:09.11 | seanjohn | I have seen that |
| 20:09.21 | seanjohn | that's what made me question qualify |
| 20:09.22 | [TK]D-Fender | seanjohn: So it plays virtually no role in the bigger scheme of failover. |
| 20:09.35 | seanjohn | I know, I was asking what did, if any |
| 20:09.41 | seanjohn | and you said nothing is available |
| 20:09.49 | [TK]D-Fender | seanjohn: it has no impact on a call taking to long once the host is considered "up" anyway |
| 20:10.38 | seanjohn | I was hoping there was a way, if the "ringing" wasn't generated within a certain ammount of time, it would hop to the next trunk; I guess I'll just have to play with different options in the dialplan to do this |
| 20:11.50 | seanjohn | like, macro and Wait(5) then gotoif($[${CONNECTED}!=1]?failovermarked or something |
| 20:12.37 | seanjohn | or ${CHANNEL(status)}=something |
| 20:12.49 | [TK]D-Fender | seanjohn: Not ever SIP service & device reports Ringing back and no, this isn't an "option" |
| 20:13.13 | [TK]D-Fender | seanAnd dialplan won't report "connected", that means you LEFT DIAL |
| 20:13.26 | [TK]D-Fender | seanjohn: I think you need caffeine. a LOT |
| 20:13.52 | WIMPy | seanjohn: *IF* you really want to do it, AMI would be your friend. |
| 20:14.19 | seanjohn | I know the dialplan won't report connected BUT ${CHANNEL(status)} or something will say RINGING and, if it does, the macro will not continue on |
| 20:14.26 | WIMPy | [TK]D-Fender: <ore likely he had too much already. |
| 20:14.35 | [TK]D-Fender | seanjohn: No, it WONT |
| 20:14.52 | seanjohn | there's no variable, automatically set, that reports the channel status of ringing? |
| 20:14.53 | [TK]D-Fender | seanjohn: Dialplan does not continue executing during your DIAL. |
| 20:15.23 | [TK]D-Fender | seanjohn: This is not happening. ANY of it. Seriously dead issue. Lights are on. The Wheel is spinning. The hampster is #&$^ING DEAD |
| 20:15.31 | seanjohn | wierd, cause it did it a couple of times when I didn't want it to, with the "g" option set |
| 20:15.39 | seanjohn | lol |
| 20:15.50 | seanjohn | burries the hamster and the hatchet that killed it |
| 20:16.08 | [TK]D-Fender | sean"G" is post dial. that means * gave up for whatever reason it chose to. This is not an opportunity for you to SUPERVISE the call |
| 20:16.40 | seanjohn | ok, thank you fender. Go take your blood pressure medication. |
| 20:24.10 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 20:24.26 | elliot98 | quietly enters and takes his seat |
| 20:24.50 | elliot98 | raises hand |
| 20:25.08 | [TK]D-Fender | is a proponent of Richard Dreyfus' "Death Therapy" method |
| 20:26.02 | elliot98 | what is the best method the dynamically change global dialplan variables? |
| 20:26.15 | elliot98 | *to |
| 20:27.17 | *** join/#asterisk eliel (~eliels@186.18.131.44) |
| 20:31.00 | [TK]D-Fender | elliot98: Meaning? |
| 20:32.27 | elliot98 | is there a way to change the value of global variable outside of a dialplan |
| 20:33.02 | elliot98 | basically, it needs to be changed, but not by a call being placed into the system |
| 20:33.08 | seanjohn | elliot, you can do it in call files |
| 20:33.38 | [TK]D-Fender | elliot98: Use AstDB instead. globals don't survive an * restart. AstDB does |
| 20:34.20 | elliot98 | and AstDB can be set using a auto-call? |
| 20:34.37 | [TK]D-Fender | elliot98: Can be set many more ways than that |
| 20:34.48 | elliot98 | I was thinking of reloading the diaplan with an updated [globals] section |
| 20:34.55 | [TK]D-Fender | elliot98: AMI, OS + * CLI, etc |
| 20:36.35 | elliot98 | gotcha |
| 20:38.37 | elliot98 | also, I've been trying to find out a way to have the background command break only if a valid digit is entered |
| 20:38.45 | elliot98 | I tried the m option, but that didn't seem to work |
| 20:39.21 | [TK]D-Fender | elliot98: It will break on any digit because SOMETHING will get hit regardless. |
| 20:39.47 | [TK]D-Fender | elliot98: thats the point of the "i" Asterisk Standard Extension. |
| 20:40.42 | elliot98 | that's what i figured...but wondering if there is some sort of way to work around that |
| 20:41.20 | elliot98 | so if an invalid digit is pressed do not stop the playback |
| 20:42.02 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 20:42.41 | [TK]D-Fender | elliot98: Nope |
| 20:44.00 | elliot98 | only voicemail playback has some playback features |
| 20:50.43 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
| 20:55.04 | *** join/#asterisk Arsenick (~rp@modemcable022.82-21-96.mc.videotron.ca) |
| 20:56.37 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
| 20:58.53 | elliot98 | so there is no way for a dialplan to break a background command only with specific digits |
| 21:00.36 | *** join/#asterisk Mango (~iMango@d154-20-93-153.bchsia.telus.net) |
| 21:02.12 | [TK]D-Fender | no |
| 21:05.36 | elliot98 | ControlPlayback |
| 21:08.30 | elliot98 | do ABCD dtmfs exist in Asterisk? |
| 21:09.57 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 21:11.19 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
| 21:13.33 | elliot98 | Controlplayback also breaks when a digit is entered...why? |
| 21:15.50 | boodu | hello |
| 21:20.43 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 21:26.55 | [TK]D-Fender | elliot98: Show us how you are using it |
| 21:27.13 | [TK]D-Fender | [17:08]<elliot98>do ABCD dtmfs exist in Asterisk? <- yes |
| 21:43.02 | Crymsonite | Outbound call on Server A, Dial SIP Device on Server B, I'm doing this all with AGI/AMI. I need to somehow identify on the Server B side, that the call originated from a specific DAHDI channel. Any ideas on how I can get this done? |
| 21:43.32 | Crymsonite | The DAHDI channel being the outbound call on Server A. |
| 21:51.54 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 21:58.11 | *** join/#asterisk Whtsup (~sssi@WimaxUser376-208.wateen.net) |
| 21:58.28 | *** join/#asterisk niekvlessert (~niek@82-171-252-6.ip.telfort.nl) |
| 21:58.44 | niekvlessert | hey guys! what about srtp/tls in 1.4? |
| 22:00.27 | *** join/#asterisk viq (~viq@unaffiliated/viq) |
| 22:05.06 | niekvlessert | if i do a reinvite between aastra's with srtp and tls, using asterisk 1.4, might that work? |
| 22:05.41 | niekvlessert | i guess srtp/tls only works well with trunk? |
| 22:10.39 | [TK]D-Fender | Crymsonite: Add a SIP header before you call to server B. Pick it off in server B's dialplan and do whatever you want with it |
| 22:17.01 | *** join/#asterisk cutencool (~dostnfrie@110.37.0.204) |
| 22:19.37 | niekvlessert | anyone has a nice url about libsrtp & asterisk 1.4? |
| 22:20.36 | Docteh | whats so bad about 1.6? |
| 22:22.21 | *** join/#asterisk mrchrisadams (~Adium@CPE-121-217-135-151.lnse2.cht.bigpond.net.au) |
| 22:23.27 | niekvlessert | well, the whole system is based on 1.4 :( |
| 22:23.42 | niekvlessert | nice and stable |
| 22:25.19 | niekvlessert | you have experience with srtp stuff? |
| 22:26.27 | Docteh | nope |
| 22:26.50 | niekvlessert | k |
| 22:26.59 | niekvlessert | does anyone think it's important on a lan? |
| 22:27.18 | Docteh | s means secure right? you tell me, its your lan |
| 22:28.00 | niekvlessert | well, if it's a lan, only local people can listen to calls with a sniffer... |
| 22:28.37 | niekvlessert | which can be trouble when an important person is talking... |
| 22:30.19 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 22:31.45 | Docteh | hubs or switches? |
| 22:31.46 | [TK]D-Fender | niekvlessert: 1.4 is ancient, get your platform off of it |
| 22:32.01 | niekvlessert | ok, 1.6 then |
| 22:32.07 | Docteh | also if you've got people sniffing your lan they are a bigger problem then whatever is being said over voip |
| 22:32.15 | Crymsonite | Thanks fender, didn't know about that function. |
| 22:32.19 | WIMPy | Docteh: Doesn't matter unless managed. |
| 22:32.20 | niekvlessert | how much trouble do I need to expect with 1.6 and srtp/tls? |
| 22:32.40 | niekvlessert | Docteh: agree, customer thinks otherwise..... |
| 22:35.30 | cutencool | hello |
| 22:35.41 | cutencool | can any one tell me the password of elatix a2billing ? |
| 22:36.15 | cutencool | plesh |
| 22:36.50 | cutencool | any one there? |
| 22:38.52 | [TK]D-Fender | cutencool: a2billing is not supported here, nor is elastix |
| 22:41.56 | *** join/#asterisk mharris (~mharris@fedora/mharris) |
| 22:47.38 | cutencool | k thanx. |
| 22:54.04 | *** part/#asterisk Whtsup (~sssi@WimaxUser376-208.wateen.net) |
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| 23:18.27 | *** join/#asterisk iceyp (~barry@120.136.2.5) |
| 23:18.57 | iceyp | Hey guys, does anyone know of a web based pbx like trixbox that I cna have multiple clients on, i.e. hosted pbx that we could host for clients on a single system |
| 23:20.03 | iceyp | I've trialed PBXnSIP but i find it to be a little too difficult for setting up auto attendants or IVR |
| 23:29.15 | niekvlessert | iceyp: no easy way to go, buy the asterisk book. :) |
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