| 00:00.36 | WiretapWork_ | jc319, iNum? |
| 00:01.15 | WIMPy | +700 :-) |
| 00:01.19 | jc319 | http://www.inum.net/ |
| 00:01.52 | WiretapWork_ | ah, I see |
| 00:01.56 | jc319 | When I logged on to the web control panel this option was there, apparently each account can get 1 of those for free |
| 00:13.10 | jc319 | I wonder if there are other devices like this, but cheaper http://www.voipupclose.com/latest-generation-of-wifi-ip-sip-phones |
| 00:14.59 | WIMPy | Sure, but they are more of a gadget when you're on your way, but nothing you want to use on a daily basis. |
| 00:15.45 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 00:17.13 | jc319 | I found some here http://www.voipuser.org/forum_topic_6781.html but without field test reports too expensive to try. I have a siemens phone here which tesco rebranded as dual phone (pstn+sip). it is okay but does only one single registration. I need 3 regs. the unbranded model does 3 but that one has a long bug reports thread so I was looking around. |
| 00:18.11 | WiretapWork_ | jc319, why do you need 3 when you can use feature codes and ring groups? |
| 00:18.33 | WIMPy | If you want to use it in a fixed location, stay with dect. |
| 00:19.04 | p3nguin | jc319: I have iNum. |
| 00:19.42 | WIMPy | I've also got an F1000. That's the one I ocassionally use when I'm on my way. |
| 00:19.50 | WiretapWork_ | p3nguin, is there anything you don't have it set up to do? XD |
| 00:19.56 | WIMPy | It can alt least do WPA, but not WPA2. |
| 00:20.12 | p3nguin | wiretapwork_: Meaning my PBX? Not really. |
| 00:20.20 | WiretapWork_ | lol |
| 00:20.22 | WIMPy | But the bad thing is that it won't recover if you lose signal. |
| 00:21.02 | p3nguin | wiretapwork_: Well, I don't have any out-of-the-country DIDs. Does that count for anything? |
| 00:21.07 | WiretapWork_ | lol |
| 00:21.20 | WiretapWork_ | one thing I do that you don't I guess :P |
| 00:21.32 | WIMPy | I also got two ZyXEL phones. They can only do WPA, but at least they recover when you lose signal. They will even do a seemless handover, although it dosn't exactely meet my definition of seemless. |
| 00:21.37 | WIMPy | You get a 2s pause. |
| 00:21.53 | p3nguin | I don't really see any reason to have a DID from another country right now. |
| 00:22.22 | jc319 | WiretapWork_: I don't know what is feature codecs, but I need 3 because I have 3 DIDs that I will use and I want to be able to place calls with any of them. This is for homeoffice use and I am thinking maybe I should just walk to 7960. |
| 00:22.25 | p3nguin | Maybe if I had a large clientele in a specific country then I would consider getting a number in that area. |
| 00:22.48 | WiretapWork_ | jc319, I said feature codes |
| 00:23.13 | WiretapWork_ | i.e. you dial **42x. and it dials out one DID, **43x. and it dials out another, etc |
| 00:23.22 | WiretapWork_ | you register the DIDs to asterisk and just point them at the phone |
| 00:23.46 | *** join/#asterisk celord (~celord@200.91.75.115) |
| 00:25.04 | WiretapWork_ | feature codes are obviously just a flashy term for custom dial prefixes in this case but eh |
| 00:26.47 | jc319 | It sounds good, but I have seperate line buttons on cisco deskphone I would prefer to use such a button on the wireless too if possible. Wireless phone is not a priority now but it is good to know what's a good one, I am thinking about setting up a voip system for a client of mine, like a demo to see how it goes. For example they might ask for a wireless unit, so good to know. |
| 00:27.08 | jc319 | is $60 a normal price for F1000? http://cgi.ebay.co.uk/Brand-New-UTStarcom-VoIP-SIP-WiFi-Cell-Phone-F1000-/330579643004?pt=Cell_Phones&hash=item4cf8150a7c |
| 00:27.47 | WIMPy | paid less than half of that. |
| 00:28.09 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 00:28.14 | WIMPy | The F1000G might be worth more. |
| 00:34.05 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 00:35.38 | WiretapWork_ | http://linux.slashdot.org/story/11/06/20/1826210/Court-Case-To-Test-GNU-GPL fritzbox are right off my christmas list now |
| 00:35.57 | sawgood | Are there any known disadvantages to using letters and/or 'symbols' in a SIP trunk context name (vs for example using only numbers)? |
| 00:40.58 | WIMPy | has never ben a fan of AVM, but I have to admit that their current software is not bad. The telephony part works very well. |
| 00:41.22 | WiretapWork_ | too bad theyre trying to render the GPL unenforcable |
| 00:41.27 | WiretapWork_ | as such, I'd like to see them flattened |
| 00:43.00 | WIMPy | Their support is very arrogant if you buy in less than 10000s quantities. |
| 00:43.31 | WIMPy | And that's true for 20grand products as well. |
| 00:47.32 | *** join/#asterisk Maxus2 (~Maxus2@59.191.225.49) |
| 00:49.29 | scalex000 | how to make a loop? |
| 00:49.58 | WiretapWork_ | take a piece of string and tie the ends together |
| 00:51.22 | Maxus2 | LOL i was going to say somthing similar. |
| 00:51.29 | p3nguin | sawgood: Almost no one uses "only numbers" in context names. Descriptive words make much more sense. |
| 00:52.33 | *** join/#asterisk cyborg-one (1000@188-115-172-249.broadband.tenet.odessa.UA) |
| 00:59.39 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 01:02.29 | *** join/#asterisk zarakiken (~zarakiken@219.95.80.9) |
| 01:13.25 | scalex000 | do you ahve another website like paste bin |
| 01:13.27 | scalex000 | ? |
| 01:13.38 | scalex000 | i need to share a macro not work |
| 01:13.59 | jc319 | pastebin alternative |
| 01:14.13 | p3nguin | ~pb |
| 01:14.13 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 01:16.25 | scalex000 | http://pastebin.us/7297 |
| 01:16.38 | scalex000 | pastebin.com not work |
| 01:17.29 | p3nguin | Line 4 will never be used. |
| 01:18.06 | scalex000 | ok |
| 01:18.11 | scalex000 | let me tell |
| 01:18.44 | scalex000 | I want to run the line 4 for a period of time |
| 01:18.54 | scalex000 | and then hangup |
| 01:19.08 | scalex000 | not |
| 01:19.10 | scalex000 | line 5 |
| 01:19.46 | p3nguin | http://pastebin.us/7298 |
| 01:19.55 | *** join/#asterisk rshah (~sabayonus@pc1.jmtech2-unet.ocn.ne.jp) |
| 01:21.36 | p3nguin | I have no clue what you're trying to do, but I cleaned up what I thought was obviously screwed up. |
| 01:23.53 | jc319 | OK this is how to set MOH in dialplan "exten=s,n,SetMusicOnHold(country)" but how can I do the same in SIP peer configuration? "mohinterpret=country" << is this wrong? |
| 01:24.16 | jc319 | or MusicOnHold(country) directly, maybe simpler.. |
| 01:26.38 | scalex000 | p3nguin, work but I need to add exten t |
| 01:27.08 | scalex000 | p3nguin, ? |
| 01:29.16 | p3nguin | Okay, so we just need to add it. exten => t,1,whatever() |
| 01:30.41 | p3nguin | exten => s,n,SetMusicOnHold(country) |
| 01:31.26 | p3nguin | See: mohinterpret and mohsuggest |
| 01:32.04 | *** join/#asterisk zarakiken (~zarakiken@219.95.80.9) |
| 01:34.48 | jc319 | exten => works I tried that, I also read the brief info I could find on mohinterpret. So there is no way to add a moh=<moh-class> in a peer definition? |
| 01:35.13 | p3nguin | Yes, there is. See mohinterpret and mohsuggest. |
| 01:39.55 | DrDigital | #freepbx |
| 01:39.57 | DrDigital | oops |
| 01:43.06 | zarakiken | ermmm good morning guys... |
| 01:43.25 | zarakiken | i got a beginners question... |
| 01:43.49 | zarakiken | how to determine a server spec for asterisk...? |
| 01:45.12 | p3nguin | Login. Check dmesg. |
| 01:47.02 | *** join/#asterisk joker2u (~root@li345-191.members.linode.com) |
| 01:48.20 | ChannelZ | eat cookie |
| 01:49.51 | joker2u | mixmonitor keeps breaking (I think). non systematic calls occur with no audio. Connection happens but one of the 2 parties cannot hear, or both. I 'think' it is mixmonitor but there are no errors to report. Curious if someone can suggest a place I can dig in to hopefully return with a better question? |
| 01:51.59 | joker2u | disabling mix monitor does 'appear' to fix the problem (low usage line). |
| 01:52.13 | joker2u | so it is really difficult to tell. |
| 01:52.25 | WIMPy | Is your harddisc too slow? |
| 01:52.26 | ChannelZ | Are you using TDM hardware? |
| 01:52.47 | joker2u | on linode, so it's a super puter. |
| 01:53.10 | joker2u | all voip, zero analog. |
| 01:53.17 | scalex000 | p3nguin, not work like a want, only loop, loop |
| 01:53.29 | p3nguin | So what are you trying to do? |
| 01:53.30 | ChannelZ | sounds like maybe your timing source is wanky then |
| 01:54.10 | joker2u | one sec, I'll post the timing definitions from extensions |
| 01:54.16 | ChannelZ | which on a virtual machine isn't necessarily surprising |
| 01:54.22 | scalex000 | p3nguin, I try to repeat a messages 10 seconds, http://pastebin.us/7302 |
| 01:56.34 | *** join/#asterisk Mhaddog_Mac (~anonymous@z65-50-116-231.ips.direcpath.com) |
| 01:57.48 | *** part/#asterisk Mhaddog_Mac (~anonymous@z65-50-116-231.ips.direcpath.com) |
| 01:57.52 | joker2u | http://paste2.org/p/1480996 |
| 01:58.08 | DrDigital | WIMPy, |
| 01:58.12 | joker2u | ChannelZ ? VM would do what? |
| 01:58.28 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 02:01.50 | *** join/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 02:02.03 | nny | No application 'Macro' for extension <--- why would the Macro application not load? |
| 02:02.08 | *** join/#asterisk mindCrime (~chatzilla@cpe-076-182-089-009.nc.res.rr.com) |
| 02:02.18 | WiretapWork | nny, turn verbose up |
| 02:02.22 | WiretapWork | load app_macro.so |
| 02:02.57 | nny | um |
| 02:02.59 | nny | it loaded |
| 02:03.03 | nny | and the thing shut up |
| 02:03.06 | WiretapWork | lol |
| 02:03.06 | scalex000 | p3nguin, easy or hard |
| 02:03.22 | WiretapWork | scalex000, he's probably the wrong person to ask, but most girls prefer hard |
| 02:03.31 | nny | but it's been restarted numerous times, do I have to put load macro statement in modules.conf? |
| 02:03.42 | scalex000 | lol |
| 02:03.42 | WiretapWork | nny, no, you should have autoload on |
| 02:03.49 | WiretapWork | and no have a noload |
| 02:04.00 | scalex000 | wiretapWork, :P |
| 02:04.09 | nny | WiretapWork: i do |
| 02:04.39 | nny | WiretapWork: guess I'll check the logs, could this be a module order condition? |
| 02:05.54 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 02:08.26 | scalex000 | omG why this macro not work, |
| 02:08.29 | scalex000 | http://paste2.org/p/1480996 |
| 02:08.31 | scalex000 | no |
| 02:08.39 | scalex000 | http://pastebin.us/7302 |
| 02:09.05 | *** join/#asterisk Bidik (~bidik@li267-109.members.linode.com) |
| 02:09.08 | scalex000 | how to setup timeout |
| 02:09.11 | scalex000 | ? |
| 02:09.19 | scalex000 | i put 10 sec |
| 02:09.31 | scalex000 | but delay 60 sec |
| 02:09.33 | scalex000 | or more |
| 02:10.37 | *** join/#asterisk timholum (~tholum@75-121-151-209.dyn.centurytel.net) |
| 02:10.41 | scalex000 | any idea :D |
| 02:14.35 | timholum | Is there a way to find out if a sip line is on the phone or not using ajam |
| 02:15.14 | nny | well this is stupid |
| 02:16.17 | nny | this is all that ERROR reports on a load, but app_macro.so isn't loading and I can't make calls. restarted today, trying to figure out who f&*#@d up what |
| 02:16.36 | nny | i can load the macro, but that's not resolving other issues |
| 02:18.26 | scalex000 | help, http://pastebin.us/7304 |
| 02:24.07 | *** join/#asterisk Kumbang (~unknown@180.245.137.5) |
| 02:26.31 | WIMPy | DrDigital: pong |
| 02:28.18 | scalex000 | ;( |
| 02:29.56 | *** join/#asterisk CaptainPants (~CaptainPa@nat/digium/x-fkmgfqlagblrfezu) |
| 02:34.21 | nny | ok i have narrowed the issue down, seems my codec_g729a.so module from digium is breaking a bunch of things on load |
| 02:34.35 | WiretapWork | lol |
| 02:36.34 | nny | WiretapWork: oh I agree. So they added a transcoder card, and sangoma loaded the drivers for it. I removed any sangoma modules or conf files, so it's not trying to load those, is there a chance the presence of this card with loaded drivers is breaking asterisk's native g729 codec? |
| 02:37.01 | nny | WiretapWork: this is from the digium repo, i can redownload and rerun the benchmark etc, but this seems to be some kind of conflict manifesting itself in horrid ways |
| 02:38.25 | nny | WiretapWork: I had to reregister the license today, but nothing in the log states it's the cause and usually g729 show licenses would just say "no" if it was incorrect, none of this breaking every other module stuff |
| 02:38.39 | scalex000 | http://pastebin.us/7304 |
| 02:40.02 | nny | wait |
| 02:40.06 | nny | hmm |
| 02:40.53 | nny | wow |
| 02:40.55 | nny | i swear to shit |
| 02:41.15 | nny | they should have to rent hands to touch these things sometimes, and pass a test first |
| 02:43.12 | WIMPy | ~collectdebug |
| 02:43.12 | infobot | extra, extra, read all about it, collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 02:43.19 | WIMPy | scalex000: ^^ |
| 02:44.24 | WIMPy | Is the line with the Swift call truncated by PB? |
| 02:45.10 | nny | welp, they also broke freepbx with this change.. son of a bitch |
| 02:45.37 | nny | hate :D |
| 02:45.46 | WIMPy | nny: What? |
| 02:47.00 | nny | WIMPy: so sangoma logged into my system today to install a transcoder and did yum -y update |
| 02:47.09 | nny | WIMPy: well one of the systems I manage |
| 02:47.26 | WiretapWork | wow....... |
| 02:47.31 | WiretapWork | that's not cool |
| 02:47.41 | nny | WIMPy: so now asterisk is 1.8.4.2, which this version of fpbx doesn't support. I know this isn't fpbx channel, (I deal with both) but i needed to rant :\ |
| 02:48.25 | *** join/#asterisk freakazoid0223 (~matt@pool-173-49-209-91.phlapa.fios.verizon.net) |
| 02:49.12 | WIMPy | wonders what packages are good for if the dependencies aren't complete. |
| 02:50.34 | WIMPy | BTW: If you install from source, that can't happen. :-) |
| 02:50.57 | nny | WIMPy: I would say they'd be fine without freepbx. Oh don't need to preach to me brother, I <3 source. However I didn't make the rules here.. :S |
| 02:51.02 | p3nguin | Don't forget to bill Sangoma for all the hours they cost you by updating something without permission to do so. |
| 02:51.19 | WiretapWork | ^ |
| 02:51.25 | nny | p3nguin: oh, I'll be interested to see the outcome of this, I just get to clean up the shit after the horses charged though |
| 02:51.27 | nny | through* |
| 02:51.46 | nny | need to see how I can specify package versions from digium's repo |
| 02:52.04 | p3nguin | Download them manually and install with rpm. |
| 02:52.14 | nny | p3nguin: does it come with a gun?! |
| 02:52.18 | nny | p3nguin: :D |
| 02:52.37 | nny | not to mention what other dependencies may have been updated but aren't compatible.. grrrr |
| 02:53.38 | nny | it's that or update freepbx, the former seems less painful and more likely to get things back to normal before I start.. ok off to manually install 8 packages and resist the urge to drink myself to death.. thanks all |
| 02:55.30 | p3nguin | So I was looking at this TRENDnet access point, and was curious how I can tell if I can use it as a "client" rather than just an access point (master) node. I'd like to hook a good antenna to it and link to a somewhat distant hotspot. |
| 02:55.47 | nny | anyone know how to browse the digium repos manually? |
| 02:55.55 | p3nguin | http |
| 02:56.08 | nny | i feel stupid, but the downloads.asterisk.org doesn't seem to list them.. |
| 02:56.08 | nny | just using the repo addy? |
| 02:56.10 | nny | k |
| 02:56.40 | p3nguin | Maybe this is a clue: "The TEW-430APB also provides WDS bridging functions, allowing users to connect two or more Access Points together." |
| 02:57.01 | WiretapWork | nope |
| 02:57.04 | WiretapWork | WDS is not client mode |
| 02:57.15 | WiretapWork | WDS is a repeater type configuration |
| 02:57.43 | p3nguin | I'd like to connect the LAN jack to a computer's Ethernet and use the device like a bridge. |
| 02:57.54 | p3nguin | WDS doesn't allow that? |
| 02:58.03 | WIMPy | If you want client you either need some advanced router like some microtic thing or something that can run openwrt or ddwrt. |
| 02:58.16 | WiretapWork | p3nguin, not properly |
| 02:58.17 | WIMPy | As WiretapWork said. |
| 02:58.17 | nny | wrt54gl works well |
| 02:58.36 | nny | and would bridge point to multi point with WDS |
| 02:58.44 | nny | well with ddwrt or tomato etc |
| 02:59.02 | WiretapWork | nny, WRT54GLs are too expensive for p3nguin :P |
| 02:59.27 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-212.ks.ks.cox.net) |
| 02:59.47 | scalex000 | too long |
| 02:59.48 | p3nguin | They cost about twice as much as I want to invest in this project. |
| 03:00.20 | nny | p3nguin: ahh gotcha, I was gettin em for 20 bucks used back when the cable guys around here needed 20 bucks lol |
| 03:00.39 | p3nguin | I'll take one for $20. |
| 03:02.23 | p3nguin | Maybe I can get a refurbished one and keep it within the low budget. |
| 03:03.24 | nny | http://www.buy.com/prod/cisco-linksys-wrt54gl-wireless-g-broadband-router-refurbished/q/loc/101/217467977.html |
| 03:03.47 | nny | check the firmware version etc, to make sure it's compatible with ddwrt etc |
| 03:04.10 | nny | may also be cheaper routers out there |
| 03:04.24 | nny | the ddwrt compatibility list is good, prob some discussions on price points on their forums too |
| 03:05.14 | p3nguin | The WRT54GS is out, right? No good for the alternate firmware, I suspect. |
| 03:06.06 | p3nguin | I don't really have any way to check the firmware version installed until I buy it and get it hooked up. |
| 03:06.54 | *** join/#asterisk scalex000 (~chatzilla@186.6.142.76) |
| 03:08.56 | p3nguin | Hmm, the WRT54GS is on their list of supported products. |
| 03:10.13 | WiretapWork | only earlier versions |
| 03:12.50 | *** join/#asterisk scalex000 (~chatzilla@186.6.142.76) |
| 03:16.15 | p3nguin | If you think I'll have better luck with the L version, I'll probably get one of those refurbs. |
| 03:18.29 | p3nguin | I just wonder how I can find out before buying if it has detachable antennas rather than the stupid molded ones. That would really piss me off to not be able to take off the antennas and connect my coax. |
| 03:18.51 | WiretapWork | the L is for Linux |
| 03:18.56 | scalex000 | ok |
| 03:18.58 | scalex000 | im back |
| 03:19.08 | scalex000 | I think here is the problem http://pastebin.us/7308 |
| 03:19.10 | WiretapWork | and WRT54G* all have RP-TNC mounted antennas |
| 03:20.00 | p3nguin | no they don't. |
| 03:20.30 | p3nguin | According to a report on the ddwrt web site, there was a period where the L model did not have removable antennas. |
| 03:21.27 | p3nguin | "it has been reported as of September 26, 2009, that for up to a year and a half ago until the present, the version of the WRT54GL you have acquired in that time period may have molded antennas which are NOT REMOVABLE" |
| 03:21.57 | p3nguin | If I get a refurb, there is no way to know when it was made. |
| 03:22.14 | p3nguin | So I'll be taking a chance. |
| 03:22.34 | p3nguin | But buy.com has a 45-day return, so not a huge issue I guess. |
| 03:23.34 | WiretapWork | p3nguin, I think that would just be idiots who don't know how to remove them |
| 03:23.41 | WiretapWork | I own four of them bought over a five year period |
| 03:23.49 | WiretapWork | not one has molded antennas |
| 03:24.57 | p3nguin | I've personally never seen a WRT54G with non-removable antennas, but I've also never personally seen an L model (which is the model with the report about the molded antennas). |
| 03:25.10 | p3nguin | I'll take a chance. |
| 03:25.24 | p3nguin | If it's molded on, I'll send it back. |
| 03:28.10 | p3nguin | So I'll get the router appliance, a yagi with an RP-TNC connector on the cable, and put ddwrt on it. Then I can use it as a client/bridge to link my wired computer to a remote access point? |
| 03:29.06 | WiretapWork | yep |
| 03:29.16 | WiretapWork | I recommend a GL on each end but I know youj don't want to spend that |
| 03:31.54 | *** join/#asterisk sourcode (~code@ppp-58-8-112-110.revip2.asianet.co.th) |
| 03:32.41 | *** join/#asterisk adelas (~adela@pool-173-51-116-120.lsanca.fios.verizon.net) |
| 03:36.01 | p3nguin | I have no control over the access point that's in place. I just need to provide extended range for some computers to link up with that access point. |
| 03:36.09 | *** join/#asterisk scalex000 (~chatzilla@186.6.139.250) |
| 03:37.31 | WiretapWork | ah |
| 03:37.39 | WiretapWork | you may have ack timing issues if it is too far |
| 03:39.04 | p3nguin | I honestly don't remember how far it is. It's been a while since I was there. |
| 03:39.36 | WiretapWork | anything over 1500m you need to adjust the ack timing |
| 03:39.39 | WiretapWork | on both ends |
| 03:39.48 | WiretapWork | or you'll get BSS timeouts |
| 03:40.06 | p3nguin | I think the AP is at the third building over. Maybe 500m or so. Two lots between the building where the AP is and where the computers are. |
| 03:41.43 | p3nguin | It's pretty annoying. |
| 03:42.25 | p3nguin | "Free Wi-Fi" ... if you walk over to the other building. :/ |
| 03:42.55 | *** join/#asterisk Khratos (~Khratos@66.128.60.148) |
| 03:43.03 | p3nguin | If you use the AP in the building on the other side it has a captive portal where you're required to pay. |
| 03:44.30 | p3nguin | Now I'm wondering if it's cheaper to pay for the neighbor's wifi rather than trying to use the free provided wifi that's out of range. |
| 03:44.55 | p3nguin | rather than buying the stuff to reach over to the provided services, I mean. |
| 03:52.54 | WiretapWork | lol |
| 03:55.42 | p3nguin | I've decided. |
| 03:55.48 | p3nguin | I ordered the wrt. |
| 03:57.03 | p3nguin | 'Cause even if I spend more on it than I would on the neighbor's non-free wifi, I'm going to take the wrt home with me when I'm done and use it for my Wii to eliminate an Ethernet cable running into the other room. |
| 03:57.09 | p3nguin | I hope it's going to be a win-win situation. |
| 03:57.38 | p3nguin | If not, then I've screwed up again. |
| 03:59.23 | WiretapWork | lol |
| 03:59.56 | *** join/#asterisk moniker326 (~moniker32@c-71-56-135-99.hsd1.wa.comcast.net) |
| 04:01.28 | p3nguin | Any opinion on this style of antenna? http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=270763015901 |
| 04:02.25 | p3nguin | I'm a little skeptical of the design. |
| 04:02.32 | WiretapWork | fine, a little bigger backphase than a dish, but still fine |
| 04:02.39 | WiretapWork | Yagis are tried and true antenna technology |
| 04:03.58 | p3nguin | I'll get one with a long cable and an RP-TNC so I don't have to buy cables and adapters. |
| 04:06.20 | WiretapWork | every metre of cable is approx 1dB of attenuation for a good cable |
| 04:06.28 | WiretapWork | make sure to use 50 ohm |
| 04:08.08 | p3nguin | I was looking at an antenna which has a 5m section of coax and the proper connector installed. |
| 04:08.51 | p3nguin | otherwise I'll have to get an extension cable for one of the antennas which only has a pigtail installed. |
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| 04:33.56 | p3nguin | Ah yes, the ol' "estimated delivery time of July 4 - July 20 from China" routine. I might receive the antenna before I need it. |
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| 05:01.01 | kleszcz | morning |
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| 05:07.47 | |ance|ott | hi |
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| 05:18.01 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 05:24.48 | fubada | hi |
| 05:24.59 | fubada | anyone can make a recommendation on US sip providers |
| 05:25.11 | fubada | is there a top-10 list or something |
| 05:25.22 | fubada | how to find a reputable company? |
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| 05:46.16 | kaldemar | ~itsp-usa |
| 05:46.21 | kaldemar | ~itsp |
| 05:46.22 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 05:46.35 | kaldemar | ~itsplist-us |
| 05:46.36 | 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 |
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| 06:01.44 | p3nguin | If only he would have stuck around for more than 15 minutes... |
| 06:02.07 | jsjc | I been lately having issues to connect with Skype for Asterisk⦠i thought could be other issues and didnt give much importance⦠Now that peak time starts for us skype becomes key again and its not connecting.. always user appears as logged out. How could I start to troubleshoot this issue? |
| 06:04.48 | p3nguin | Who provides your Skype for Asterisk channel driver? |
| 06:05.17 | jsjc | digium. |
| 06:06.30 | p3nguin | I'd start by emailing Digium technical support. Ask them what they recommend doing to troubleshoot. |
| 06:07.42 | jsjc | I was worried digium will wash their hands of with this thing that the end of the month they do not sell it anymore... |
| 06:07.46 | jsjc | I will give it a try |
| 06:07.51 | jsjc | Thanks for reccomendation |
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| 06:41.26 | Karen_m | I have a did that I want to have people call, but it rings 2 different cell phones. Whoever picks up first, gets connected to the call and the call is recorded in an audio that gets emailed after. Is this possible? |
| 06:41.34 | Karen_m | if it is, can someone give me a hint as to what I should google? |
| 06:47.18 | *** join/#asterisk boazb (~b@bzq-82-80-219-90.red.bezeqint.net) |
| 06:47.44 | kaldemar | yes, it is possible. in asterisk's CLI, "core show application dial" <-- [Syntax] tells how to dial 2 phones, "core show application mixmonitor" <-- records a call, "core show application system" <-- gives a system command to send e-mail. |
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| 06:54.13 | Karen_m | thank you |
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| 07:35.12 | DND | hi guys why is it when i call my mobile phone then i hang up while its still ringing, the ring on my mobile phone did not stop for a couple of seconds |
| 07:39.46 | kaldemar | please elaborate |
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| 07:54.49 | xofapcom | hi, how to route call based on DID on same gateway? |
| 07:55.58 | WiretapMac | use exten matching |
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| 07:58.19 | xofapcom | well, my gateway consist of 2 DID, and I need to separate the incoming to both DID to different extension |
| 08:04.16 | WiretapMac | so match the incoming calls |
| 08:04.25 | irroot | bitch slaps ldap ... |
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| 08:07.16 | iulhk | hi all, i need to know about voicemail, i hv confiugured it, when i press voicemail extension, i can hear voicemail IVR, like you have 5 new voicemail messenges in your inbox. if i hv 10 voicemail in my inbox 5 are new, then how can i show my un-heared voicemail messages quantity in my sip client ? or how asterisk know as IVR saying you are 5 new msgs in your inbox ? |
| 08:09.06 | *** join/#asterisk vfabi (~fabi@194.247.164.231) |
| 08:09.30 | WiretapMac | iulhk: the message count is transmitted to your endpoint (hardphone/softphone), it is up to the phone to decide what to do with it |
| 08:09.40 | irroot | iulhk notify message-summery |
| 08:10.10 | iulhk | <irroot>: in voicemail.conf ? |
| 08:10.13 | irroot | WiretapMac there is a bug in 1.8 that it does not transmit it on register as it did in 1.4 |
| 08:10.28 | iulhk | using asterisk 1.6 |
| 08:10.33 | irroot | iulhk no its a sip message sent to the device |
| 08:10.35 | WiretapMac | irroot: it sends it in the new message notify though? |
| 08:10.36 | xofapcom | WiretapMac : sorry, I'm not very familiar with asterisk dialplan. If I need to have incoming to 12345678 to go to ext 101, how should I write it in dialplan? |
| 08:11.11 | WiretapMac | xofapcom: this will help you8 http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 08:11.21 | *** join/#asterisk kaii (~kh@ciphron.de) |
| 08:11.23 | iulhk | thanx |
| 08:11.27 | *** join/#asterisk brah (~paul@host28.200-82-50.telecom.net.ar) |
| 08:11.34 | irroot | WiretapMac it does if the phone subscribes to the MWI |
| 08:11.49 | WiretapMac | mmm, most do, no? |
| 08:12.05 | irroot | iulhk https://reviewboard.asterisk.org/r/1272/ |
| 08:12.22 | irroot | wiretapmac seems if configured |
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| 08:12.37 | WiretapMac | I know my Ciscos do, don't think they say how many messages |
| 08:12.44 | irroot | the other way is to dial into voicemailmain it always generates a event |
| 08:12.44 | iulhk | let me check |
| 08:13.25 | irroot | also if a new mail comes in it will update the phone the link above causes it to be sent at register |
| 08:17.20 | kaii | guys, what is the issue here? you miss MWI on register? |
| 08:21.51 | irroot | yip kaii |
| 08:22.29 | kaii | irroot: what exactly, you subscribe but dont get the correct values? |
| 08:23.03 | kaii | irroot: see ASTERISK-17997, try this patch https://issues.asterisk.org/jira/secure/attachment/40433/asterisk-1.8.4.2-sip-mwi-event-v2.patch |
| 08:23.04 | CaptainPants | [ASTERISK-17997] [Status: Open] When subscribing mwi to an unsolicited mailbox the first notification is incorrect - https://issues.asterisk.org/jira/browse/ASTERISK-17997 |
| 08:23.06 | irroot | kaii if you subscribe for MWI explicitly it works not all devices are subscribing |
| 08:23.14 | kaii | ah i see |
| 08:23.32 | iulhk | <kaii>: my query is , i hv confiugured it, when i press voicemail extension, i can hear voicemail IVR, like you have 5 new voicemail messenges in your inbox. if i hv 10 voicemail in my inbox 5 are new, then how can i show my un-heared voicemail messages quantity in my sip client ? or how asterisk know as IVR saying you are 5 new msgs in your inbox ? |
| 08:23.34 | irroot | the old behavior worked without subscribing to MWI |
| 08:23.40 | irroot | ASTERISK-17866 |
| 08:23.41 | CaptainPants | [ASTERISK-17866] [Status: Open] [patch] MWI last-msgs-sent is mis-reported - https://issues.asterisk.org/jira/browse/ASTERISK-17866 |
| 08:24.06 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 08:25.44 | kaii | irroot: i see |
| 08:26.09 | irroot | kaii https://reviewboard.asterisk.org/r/1272/ |
| 08:32.19 | kaii | irroot: final patch (6) looks good to me |
| 08:32.52 | irroot | indeed it takes the curveballs into account |
| 08:33.25 | irroot | just need some ship it love |
| 08:34.27 | irroot | perhaps test it for ASTERISK-17997 ?? |
| 08:34.29 | CaptainPants | [ASTERISK-17997] [Status: Open] When subscribing mwi to an unsolicited mailbox the first notification is incorrect - https://issues.asterisk.org/jira/browse/ASTERISK-17997 |
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| 08:38.28 | kaii | i cant login to reviewboard with my jira user :( |
| 08:48.48 | *** join/#asterisk dror9999 (d4b38cc2@gateway/web/freenode/ip.212.179.140.194) |
| 08:48.58 | dror9999 | hi |
| 08:49.16 | xofapcom | WiretapMac : done. thanks for helping |
| 08:52.50 | tzafrir_laptop | dror9999, hi |
| 08:53.21 | tzafrir_laptop | ~ask |
| 08:53.22 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 08:57.40 | dror9999 | I'd like to monitor a call and every 20 seconds stop monitoring and start monitoring to a new file. Is there a way to do this? |
| 08:58.58 | tzafrir_laptop | Why would you want to do that? |
| 08:59.03 | kaii | dror9999: well, why do you want this? you could possibly do this with a script that controls the channel monitor using AMI |
| 08:59.52 | tzafrir_laptop | I suspect stopping and starting will get you some bits of the call missed |
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| 09:01.23 | dror9999 | kaii: I want to send the recorded file to a voice recognition service and get results back while still talking... |
| 09:02.29 | tzafrir_laptop | I suspect that there are better ways to do that |
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| 09:04.32 | tzafrir_laptop | So you don't actually need to get the call recorded? |
| 09:05.22 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 09:06.42 | dror9999 | no, I don't actualy need to get the call recorded. |
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| 09:12.19 | schmidts | good morning |
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| 10:07.06 | *** join/#asterisk MDesade (~chatzilla@ip24-251-93-137.ph.ph.cox.net) |
| 10:07.14 | MDesade | hello all |
| 10:07.43 | MDesade | i could use some help getting dahdi to work here.... |
| 10:08.14 | *** join/#asterisk jkroon (~jkroon@dsl-242-11-09.telkomadsl.co.za) |
| 10:08.37 | jkroon | hi guys, since the last update on chan_local to the 1.6.2 branch - has anybody started seeing * segfaults? |
| 10:08.50 | MDesade | its compiled installed and all that, and i can run commands just fine. strangely in the asterisk CLI> i can see the status of the hardware, yet i cant see the channels other than the psuedo dummy interface |
| 10:09.57 | jkroon | http://pastebin.com/MAFHduQ4, line 529 is the if statement that tries to test for looped back channels. |
| 10:11.23 | jkroon | secondly, i'm not receiving my password reset email on JIRA, so is there perhaps an administrator around that would be able to assist me ? |
| 10:12.25 | MDesade | http://pastebin.com/wUsN15d0 |
| 10:15.35 | tzafrir_laptop | MDesade, the channels are not defined in chan_dahdi.conf |
| 10:15.40 | jkroon | ok, three of the four core dumps actually dies in local_setoption, not in local_fixup |
| 10:15.44 | tzafrir_laptop | could you please pastebin that file? |
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| 10:16.08 | *** join/#asterisk BugKhaM (~BugKhaM@125.25.52.102.adsl.dynamic.totbb.net) |
| 10:16.16 | tzafrir_laptop | Alternatively, you can generate a working snippet using: dahdi_genconf chandahdi |
| 10:16.39 | tzafrir_laptop | It will be generated in /etc/asterisk/dahdi-channels.conf |
| 10:16.46 | tzafrir_laptop | Add it to the end of chan_dahdi.conf |
| 10:17.02 | *** join/#asterisk ltd (~z@zx.io) |
| 10:17.09 | jkroon | tzafrir_laptop, i don't suppose you can help me to reset my password on issues.asterisk.org? |
| 10:17.36 | MDesade | tzafrir chan_dahdi.conf ? |
| 10:18.13 | MDesade | i DO have an "include" in that file, that points to dahd-channels.conf |
| 10:18.15 | BugKhaM | Hi, I'm using the ' L' option with "cmd Dial()". The problem is that the 'L' option won't work if it's used within another Dial(). |
| 10:18.45 | BugKhaM | like if I'm using Dial(SIP/) within Dial(Local/) |
| 10:19.00 | BugKhaM | will Set(TIMEOUT(absolute)=value) |
| 10:19.11 | BugKhaM | work for this? |
| 10:20.19 | kaldemar | BugKhaM: if you use /n with the local channel, it might work. |
| 10:24.03 | tzafrir_laptop | where did jkronn go? |
| 10:28.45 | BugKhaM | kaldemar: will give it a try, what does the "/n" do by the way? |
| 10:30.00 | BugKhaM | kaldemar: okay, found a link about that now |
| 10:30.23 | MDesade | http://pastebin.com/ySnBFuHK |
| 10:37.22 | MDesade | here is what i see in the CLI http://pastebin.com/rnQyxMmx |
| 10:41.22 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 10:41.22 | *** mode/#asterisk [+o file] by ChanServ |
| 10:43.32 | MDesade | here is dahdi-channels.conf http://pastebin.com/TexWiXn5 |
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| 11:27.11 | hrolf | In the 'stream file' command, if it returns 'endpos=0' should I take that as an error? |
| 11:29.22 | irroot | is seen lots of problems with codec negotiation |
| 11:31.21 | kaldemar | MDesade: "include = /etc/asterisk/dahdi-channels.conf" should be "#include /etc/asterisk/dahdi-channels.conf" |
| 11:32.26 | hrolf | How do you navigate asterisk's source code? How can I import it in Visual Studio or Eclipse, so that I may be able to run a search function etc? |
| 11:33.03 | irroot | hrolf use vi ?? :P |
| 11:33.28 | kaldemar | hrolf: are you familiar with grep? |
| 11:33.33 | irroot | should be able to use eclipse quite easy |
| 11:33.37 | hrolf | irroot: there is learning involved :) |
| 11:33.48 | hrolf | kaldemar: yes, that's what I'm using right now |
| 11:33.49 | WIMPy | You could use the doxygend stuff, gut I found grep more effective. |
| 11:34.22 | irroot | confesses to using google to search doxy bits |
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| 11:36.49 | kaii | hrolf: with the exclipse CDT extension it is quite easy to import asterisk as a project from existing sources. i do not understand your question. did you already try it on your own? |
| 11:37.12 | hrolf | kaii: no I haven't tried it in Eclipse |
| 11:37.38 | kaii | dont know about MS foo, but eclipse and netbeans were both quite easy to use on asterisk sources |
| 11:38.19 | kaii | hrolf: btw, what irroot wants to tell you: the source documentation (doxygen) will help you even better than any IDE can do. |
| 11:38.42 | hrolf | actually I'm on Windows and just wanted to read asterisk's source code, downloading Eclipse now |
| 11:38.54 | tuxx- | eclipse to read asterisk source code? |
| 11:38.55 | kaii | you can read it online |
| 11:39.01 | kaii | and with notepad |
| 11:39.03 | kaii | or word |
| 11:39.09 | kaii | or whatever you have on windows |
| 11:39.12 | hrolf | kaii: you can't search with it |
| 11:39.23 | kaii | hit CTRL^F |
| 11:39.27 | tuxx- | vim file.c |
| 11:39.31 | tuxx- | <PROTECTED> |
| 11:39.31 | hrolf | kaii: in one file only |
| 11:39.31 | tuxx- | :D |
| 11:39.39 | tuxx- | :split 2ndfile |
| 11:39.42 | tuxx- | \o/ |
| 11:39.53 | tuxx- | fgrep -ir string . |
| 11:39.56 | tuxx- | etc. etc. |
| 11:39.57 | tuxx- | :D |
| 11:40.02 | kaii | hrolf: as said, if you want to understand the source, read the source docs. |
| 11:40.26 | hrolf | tuxx-: If eclipse can load all of the asterisk project, then it is good. I can just hit Go to Definition and I'll be there |
| 11:40.42 | tuxx- | :P |
| 11:40.53 | tuxx- | yeah i dont like eclipse all too much |
| 11:40.56 | tuxx- | just like any other IDE |
| 11:40.57 | hrolf | tuxx-: can grep do tha? |
| 11:41.00 | hrolf | that* |
| 11:41.09 | kaii | hrolf: http://www.asterisk.org/developers/documentation |
| 11:41.12 | tuxx- | grep can display the file your searchterm is in |
| 11:41.16 | tuxx- | and you can edit the file with vim :P |
| 11:45.54 | hrolf | tuxx-: grep will give you the files where it is referenced, and you'll have to look in all files to look for its definition |
| 11:45.57 | hrolf | tuxx-: : ) |
| 11:46.41 | MDesade | kaldemar - thank you, you are correct, fixed and working |
| 11:48.01 | tuxx- | hrolf: then ur not grepping right ;-) |
| 11:48.37 | hrolf | tuxx-: because I do not know regex :) |
| 11:49.00 | hrolf | tuxx-: what will be the regex to find the definition of ast_waitstream_full |
| 11:49.33 | hrolf | tuxx-: ast_waitstream_full(*){ in wild characters, how to convert it into regex? |
| 11:49.37 | tuxx- | you dont have to use regex's |
| 11:49.47 | tuxx- | could also use taglist ;-) |
| 11:50.00 | tuxx- | vim taglist that is |
| 11:50.30 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:50.30 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:50.31 | hrolf | tuxx-: i'm using grep |
| 11:50.51 | hrolf | tuxx-: like ast_waitstream_full(<the regex to match any character) { |
| 11:51.08 | hrolf | tuxx-: what will be the <the regex to match any character(s)> |
| 11:51.29 | Jarris | i do need some help with AsteriskNOW - Asterisk GUI, since there is no other language available, but english, i need a workaround e.g. an german tooltip.js. can anybody help me? thanks in advance |
| 11:53.26 | irroot | coffee indicator is at dangerous low feeling excessively misanthropic |
| 11:55.38 | tuxx- | hrolf: something like ast_waitstream_full\(.*){ |
| 11:56.29 | hrolf | tuxx-: invalid regex |
| 11:57.00 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 11:57.43 | tuxx- | is this a test or something? :D |
| 11:57.48 | tuxx- | where can i redeem my price. |
| 11:58.25 | WIMPy | Jarris: This channel is not very GUI friendly. But I guess you could try Gemeinschaft. That probably has a german version. |
| 11:59.33 | irroot | volenteers are sought to do some translation ??? |
| 11:59.56 | hrolf | tuxx-: It ain |
| 12:00.04 | hrolf | tuxx-: * It ain't one :) |
| 12:00.07 | tuxx- | hehe |
| 12:00.10 | hrolf | tuxx-: I'll find it out my self thanks :) |
| 12:00.19 | kaldemar | Jarris: see if there are people on #asterisknow that might be able to help. |
| 12:01.20 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 12:02.25 | Jarris | kaldemar: thanks WIMPy: i dont have the choise, i have to convert this one in germen, thanks anyway |
| 12:06.45 | Faustov | ~ask |
| 12:06.45 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:11.00 | adyn | ~book |
| 12:11.00 | 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://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 12:12.24 | cneb3000 | ~cheese sandwich |
| 12:12.24 | infobot | ACTION hands sandwich a tray of sliced sharp cheddar and pineapple |
| 12:12.34 | cneb3000 | (had no idea it would do that) |
| 12:12.48 | *** join/#asterisk fish-bulb (~qcstewart@nat/digium/x-woeyavetjasyqgdc) |
| 12:12.48 | adyn | I had no idea it included pineapple... |
| 12:12.56 | tuxx- | cheddar != cheese |
| 12:13.00 | adyn | not sure that's a good combo with sharp cheddar |
| 12:13.31 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:13.44 | cneb3000 | ~beer |
| 12:13.44 | infobot | ACTION has disconnected (Read error: 99 (Connection reset by beer)) |
| 12:13.52 | cneb3000 | :| |
| 12:13.55 | tuxx- | heh |
| 12:14.05 | cneb3000 | oh wait |
| 12:14.15 | cneb3000 | sure got me. |
| 12:14.20 | cneb3000 | lol |
| 12:14.23 | tuxx- | oh you thought he actually dc'ed? |
| 12:14.24 | tuxx- | ;D |
| 12:14.29 | tuxx- | hehehe |
| 12:14.30 | cneb3000 | ...............no ¬_¬' |
| 12:14.33 | tuxx- | ;-) |
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| 12:21.00 | irroot | lo there cneb3000 hehe |
| 12:21.08 | cneb3000 | ...dont look at me ¬_¬ |
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| 12:25.24 | irroot | geez someone actually left me a voicemail .... poor them well +1 experiance |
| 12:26.15 | mha_ | just installed asteriskNOW, but donno the user and password |
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| 12:29.44 | mha_ | anyone knows user/password for asteriskNOW GUI? the default |
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| 12:45.38 | madmax78 | anybody familiar with xzorcoms asterisk patch to record calls to remote server .eg oreka |
| 12:45.57 | mha_ | anyone knows user/password for asteriskNOW GUI? the default |
| 12:47.40 | tuxx- | http://www.lmgtfy.com/?q=asterisknow+default+password |
| 12:53.52 | *** join/#asterisk vfabi (~fabi@194.247.164.231) |
| 12:59.47 | DND | hi guys anyone know any good call logging software that will just log the call and log the amount it took? |
| 13:00.08 | Tozz_ | cdr log? |
| 13:00.13 | DND | a2billing is out of the question because its too complex and has too many feature |
| 13:00.20 | otwieracz | amount of what? |
| 13:00.29 | DND | Tozz_, yes but with a call cost feature |
| 13:00.43 | Tozz_ | ah |
| 13:01.05 | DND | but i only need reports. not billing |
| 13:04.12 | irroot | CDR -> SQL is a good win |
| 13:04.20 | irroot | can do simple reports |
| 13:04.58 | cneb3000-away | ^^ yea, the length of the call is probably already being written. a query/stored procedure could probably be written to calculate 'cost' too |
| 13:05.16 | Katty | nomnombreakfastnomnom |
| 13:05.56 | irroot | cneb3000-away yeah have a "billing" table with the telco costs |
| 13:06.10 | irroot | does the costs and the works |
| 13:06.46 | cneb3000 | DND: any good with SQL? :D |
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| 13:17.47 | irroot | katatonic <- happens when you chow as much sugar as her |
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| 13:22.34 | Katatonic | i didn't chow sugar |
| 13:22.41 | Katatonic | but i did eat breakfast. |
| 13:22.44 | Katatonic | still hungry tho )= |
| 13:22.59 | Katatonic | all that biking is catching up with me |
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| 13:33.01 | gopal_ | what is the reason for asterisk dead subsys locked? |
| 13:33.17 | Katatonic | i think i might be getting the ills. |
| 13:34.02 | *** join/#asterisk b0ot (~Jinxed---@147.177.56.129) |
| 13:34.11 | b0ot | I want to start collecting all my asterisk/cisco configs (and other phone configs) and make them searchable in some fashion. Anyone know of any good tools for that? |
| 13:34.24 | Katatonic | wordpress |
| 13:35.29 | WIMPy | grep |
| 13:36.04 | gopal_ | what is the reason for "asterisk dead subsys locked" when i do "service asterisk status"? |
| 13:36.36 | b0ot | WIMPy, Katatonic I'm looking more of something where I could tag configs with things such as the type of phone, brief descriptions, etc and then search on those terms as well |
| 13:36.48 | WIMPy | gopal_: I guess you need to ask that to who ever made "service". |
| 13:36.57 | irroot | bOot Joomla ?? |
| 13:37.28 | gopal_ | WIMPy: from the asterisk source i did make config... so that it will installed as service... |
| 13:37.39 | WIMPy | b0ot: You can place comments in all config file. Or you do indeed use any sort of CMS/Wiki/... |
| 13:37.54 | Katatonic | blogger and wordpress have tags |
| 13:38.17 | WIMPy | gopal_: I think Windows has a concept of "services". What are you talking about? |
| 13:38.56 | gopal_ | WIMPy: as a linux service when we do "make config" from the asterisk source folder asterisk will run as background service.... |
| 13:39.24 | WIMPy | We use to call them daemons. |
| 13:40.00 | WIMPy | Anyway I have no idea what kind of script or concept it's about. |
| 13:40.02 | gopal_ | WIMPy: ok then my lines might be wrong... |
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| 13:40.30 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:41.48 | gopal_ | WIMPy: now suddenly the daemon is running i am able to check with "service asterisk status" also able to enter into client mode "asterisk -r" I dont know what was wrong |
| 13:42.24 | WIMPy | Ok, glad it works. |
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| 13:46.39 | irroot | is a narcissistic SOB .... got the whole office trying to find the auto-prov sequence of yealink phone im not arsed to figgure it out |
| 13:49.17 | whirlpbx | Hello, I have three IP-phones, that have to ring on incoming call from FXO-line. Two of them have to start ringing immediately and third after 20 seconds. Any ideas? |
| 13:50.14 | WIMPy | whirlpbx: Create an extension for the 3rd phone with a Wait() before the dial and dial that with the other two via a local channel. |
| 13:50.39 | WIMPy | If you can't tell your phone to wait, that is. |
| 13:56.18 | whirlpbx | WIMPy: Dial(SIP/21&SIP/22&LOCAL/23@custom_context)? And put extention 23 with Wait(20) in "custom_content"? |
| 13:56.45 | whirlpbx | *custom_conteXt |
| 13:57.22 | WIMPy | Something like that, yes. |
| 13:57.40 | whirlpbx | thx, i'll try. |
| 13:57.51 | WIMPy | You could even buid a generic delay extension. |
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| 14:24.32 | mboylan | Hi guys -- anyone around? Having a SIP problem that I can't seem to get fixed |
| 14:24.42 | WIMPy | ~ask |
| 14:24.42 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:25.17 | cneb3000 | mboylan: yes people are around :) |
| 14:25.44 | WIMPy | And some are asquare. |
| 14:26.18 | cneb3000 | how can you be asquare :| |
| 14:26.56 | WIMPy | In a cubicle perhaps. |
| 14:27.04 | mboylan | We're finally making the move from 1.2 Business Edition to 1.8 Open Source. We're replacing the two asterisk servers at the routing layer first. Boss wants to move away from IAX and go all SIP. I have sip trunks made between all of the asterisk servers. When trying to pass the call from the 1.8 server to a 1.2 server, I'm getting the dreaded "No reply to our critical packet" problem. IPTables is turned off on both servers and, for testing purposes, we've |
| 14:27.04 | mboylan | the 1.8 machine into a non-firewalled subnet |
| 14:27.29 | cneb3000 | WIMPy: touche |
| 14:28.15 | cneb3000 | mboylan: Have you made any configuration changes in sip.conf in order to get this working? |
| 14:28.37 | mboylan | On the 1.2 server whenever a call comes in from the 1.8 server, the message -- Incoming call: Got SIP response 491 "Request Pending" back from 10.96.192.14 is shown over and over |
| 14:28.47 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:28.50 | mboylan | the 10.96 IP is the 1.8 server |
| 14:29.13 | mboylan | cneb3000: I've tried a few things but mostly just trying everything I could find on resolving the critical packet issue. |
| 14:29.25 | mboylan | If I keep asterisk in the middle (no reinvites), it works fine |
| 14:30.04 | mboylan | I'm going to be working with our lead networking engineer later to do some more network tracing, but ports are open and there's no NAT. There really shouldn't be too much magic involved here |
| 14:31.10 | mboylan | the entries in SIP.conf on both sides looks like this: http://pastie.org/private/a63snlrjzzw9qfdxm5hzqa |
| 14:31.13 | cneb3000 | so you're passing a call from asterisk 1.8 to asterisk 1.2? when you say 'passing'... do you mean transfer or? |
| 14:32.04 | mboylan | the polycom phones are registered on Asterisk5 (1.2). In the dialplan on Asterisk14(1.8), the _69xx extensions dial SIP/Asterisk5/${EXTEN} |
| 14:32.43 | mboylan | I have some sip debugging output from the 1.8 box if it would help? :) |
| 14:33.48 | mboylan | http://pastie.org/private/rzencbfkd7lem7br9ntw |
| 14:34.02 | scalex000 | good morning |
| 14:35.19 | mboylan | cneb3000: On the 1.8 side I've tried some of the newer rtp options but they didn't seem to make a difference. I can't touch much on the 1.2 side as it's in production |
| 14:35.55 | cneb3000 | why are you doing it between 1.2 and 1.8? cant you do it between 1.8 and 1.8? |
| 14:35.59 | cneb3000 | or is this all just for testing? |
| 14:36.42 | mboylan | cneb3000: because our asterisk environment is distributed. We have 5 servers... 2 of which phones register two and two which do the routing/connect to the trunks, etc. We can't just roll out everything at once |
| 14:37.05 | mboylan | The 5th one is the polycom ftp server and sort of serves as a third failover for everything else |
| 14:37.32 | cneb3000 | out of my depth unfortunatly... hope someone else in the room can pitch in.. |
| 14:37.35 | mboylan | We wanted to roll out 1.8 at the routing layer and then do the other two boxes soon thereafter |
| 14:38.09 | mboylan | But just doing it all at once one night and hoping it all works the next day is a bit too risky, haha :p |
| 14:38.30 | mboylan | k, no problem. Appreciate you listening anyhow :) |
| 14:38.54 | mboylan | Anyone else around have any thoughts? |
| 14:39.20 | *** join/#asterisk phingers (~chatzilla@173-162-124-105-miami.hfc.comcastbusiness.net) |
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| 14:40.13 | WIMPy | From your description it seems to be a networking issue, but I haven't looked closes as I try to avoid SIP if any possible. |
| 14:40.15 | phingers | having troubles with of course, the bosses phone |
| 14:40.30 | mboylan | WIMPy: Do you use IAX instead? |
| 14:40.33 | leroybuckingham | Hey is there any reason to choose 32-bit over 64-bit installations with asterisk? |
| 14:40.55 | phingers | we have it ring one phone, then follow me to a second phone in his office. Problem is his main phone will ring once, then stop, and only ring in the office the rest of the time. |
| 14:41.09 | phingers | it happens randomly |
| 14:41.16 | phingers | sometimes it works just fine |
| 14:41.30 | WIMPy | mboylan: Between Asterisks I use IAX, yes. |
| 14:42.15 | mboylan | We use it now. not sure why the boss wants to move away from it. If I can't get this SIP working w/ 1.2 and 1/8,I may try IAX again as a workaround until 1.8 is fully deployed |
| 14:42.20 | phingers | so - any recommendations on a large asterisk support company that can help me? |
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| 15:15.02 | Katty | cranks the heat up |
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| 15:24.04 | scalex000 | hello |
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| 15:24.59 | wwalker | VOIP provider just told a client "If your gateway is asterisk we will not be able to support your issue due to known issues with Asterisk and its DTMF method". Anyone got any idea what crap he's talking about so I can find that it is fixed and refute it? |
| 15:25.52 | scalex000 | I need help how to call another ext http://pastebin.com/yYZKpjzD |
| 15:28.07 | p3nguin | Just pick up your phone and dial the numbers. That's how you call an extension. |
| 15:28.52 | *** join/#asterisk wonderworld (~ww@port-92-201-198-82.dynamic.qsc.de) |
| 15:29.00 | kaii | scalex000: "core show application Goto" ? |
| 15:29.08 | kaii | (type that on your asterisk CLI) |
| 15:29.51 | scalex000 | ok |
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| 15:44.34 | Katty | weeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeee |
| 15:48.19 | *** join/#asterisk wonko (~wonko@wiggum.4amlunch.net) |
| 15:48.33 | wonko | anyone have a working link for AsteriskNOW 1.7.1? |
| 15:48.41 | wonko | the links on the asterisk website give me a 404 |
| 15:56.42 | serafie | wonko: http://www.asterisk.org/downloads/asterisknow/i386/asterisknow32.iso |
| 15:57.46 | wonko | awesome, thanks! |
| 15:58.02 | serafie | yw! |
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| 16:12.17 | Qwell | wonko: which links? from where? |
| 16:14.53 | *** join/#asterisk suzie_needs_help (48edd5a2@gateway/web/freenode/ip.72.237.213.162) |
| 16:16.59 | suzie_needs_help | we've been getting complaints that DTMF is not working, i checked full log for a call that they claimed DTMF didnt work, I noticed this: DTMF begin '7' received, DTMF begin ignored, DTMF end '7' received, DTMF end passthrough '7' why is the DTMF begin ignored called? |
| 16:18.12 | wonko | Qwell: the links on http://www.asterisk.org/asterisknow/ to download |
| 16:18.54 | Qwell | hmm |
| 16:18.57 | Qwell | thehar: YOU BROKE IT |
| 16:19.14 | Qwell | thehar: actually you didn't, but any idea who handles the mirrors at xmission? |
| 16:19.22 | thehar | i broke it? |
| 16:19.25 | thehar | yes |
| 16:19.33 | thehar | Qwell: join irc.xmission.com and talk to josh or pashdown |
| 16:19.35 | Qwell | asterisk mirror is apparently 404 :( |
| 16:19.35 | thehar | or logik |
| 16:19.39 | thehar | or go to livesupport.xmission.com |
| 16:19.43 | thehar | ah.. didn't tell them to rsync :) |
| 16:19.52 | thehar | or changed a path on them i bet |
| 16:19.52 | thehar | =P |
| 16:19.56 | Qwell | the whole dir got removed |
| 16:20.01 | thehar | oh wow |
| 16:20.03 | thehar | here i'll poke someone |
| 16:20.06 | Qwell | no more asterisk mirror at all :( |
| 16:20.17 | thehar | i'm poking them |
| 16:20.35 | Qwell | w00t |
| 16:20.37 | p3nguin | wonko: Did you get the file you need? I have AsteriskNOW-1.7.1-i386.iso if you need it. |
| 16:20.51 | Qwell | p3nguin: sounded like he did |
| 16:21.07 | wonko | yeah, serafie got it to me |
| 16:21.18 | Qwell | which is funny, because she gave the same link that's on the website. heh |
| 16:21.25 | Qwell | silly roundrobin mirror |
| 16:21.36 | thehar | hehe |
| 16:21.38 | wonko | ha |
| 16:21.41 | thehar | refresh repeatedly |
| 16:21.44 | wonko | i reloaded the page and now i get those links |
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| 16:23.15 | irroot | got a #physics question tonight been the longest night i get too sleep in right ?? |
| 16:23.23 | thehar | Qwell: hit me up in social or query |
| 16:23.26 | suzie_needs_help | we've been getting complaints that DTMF is not working, i checked full log for a call that they claimed DTMF didnt work, I noticed this: DTMF begin '7' received, DTMF begin ignored, DTMF end '7' received, DTMF end passthrough '7' why is the DTMF begin ignored called? |
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| 16:33.50 | p3nguin | That mirror has been screwed up since at least when I discussed it with someone weeks ago. |
| 16:43.41 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
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| 16:50.23 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 16:58.08 | weinerk | s |
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| 17:00.21 | *** join/#asterisk rutski (~rutski@96.56.54.186) |
| 17:00.24 | rutski | hey there |
| 17:00.30 | *** join/#asterisk weinerk (~user@unaffiliated/weinerk) |
| 17:00.50 | rutski | where can I get detailed documentation about the options in a given asterisk config file? |
| 17:01.08 | rutski | I need to know what the whole range of options is, what exactly they do, what their defaults are, etc... |
| 17:01.37 | Qwell | sample config file |
| 17:01.53 | rutski | hmm, really? |
| 17:02.02 | rutski | there's nothing like, for example, what apache has for a given module: kind of like this sort of thing for, for example, |
| 17:02.04 | rutski | http://httpd.apache.org/docs/2.2/mod/mod_mime.html |
| 17:02.19 | rutski | (type-o there on that line before the link, sorry) |
| 17:02.57 | atheos_ | rutski voip-info.org has a great deal of information, also from the CLI: show application ... |
| 17:03.11 | rutski | yeah, the "show application" thing has been awesome :D |
| 17:03.37 | atheos_ | rutski also, from the asterisk source directory, doc/AST.pdf |
| 17:03.57 | rutski | yeah, about that |
| 17:04.01 | rutski | either my PDF reader is really broken |
| 17:04.04 | rutski | or that PDF file is really broken |
| 17:04.16 | rutski | because all of the "code boxes" are cut off or messed up in some way |
| 17:04.18 | atheos_ | It's also there in .txt |
| 17:05.05 | rutski | doesn't look very readable |
| 17:05.24 | rutski | this is appealing though: http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf |
| 17:05.27 | atheos_ | http://www.amazon.com/Asterisk-Dummies-Stephen-P-Olejniczak/dp/0470098546 |
| 17:05.59 | *** join/#asterisk hdiogenes (~humberto@201.76.154.125.intranet.digi.com.br) |
| 17:06.33 | suzie_needs_help | does "DTMF begin ignored" mean asterisk ignored the DTMF signal? |
| 17:06.50 | rutski | atheos_: wait... a sec, do you see on http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf where it documents the "Predefined Extension Names"? |
| 17:07.05 | rutski | like "s" for "start" and "T" for AbsoluteTimeout |
| 17:07.23 | rutski | is voip-info.org *really* the only place to find that sort of information about a config file? |
| 17:07.25 | atheos_ | Yes rutski, what about them |
| 17:07.44 | atheos_ | No, I'm pretty sure that is all detailed in the AST.pdf file |
| 17:08.15 | rutski | atheos_: (just to double check, the AST file is the same as https://wiki.asterisk.org/wiki/display/AST/Asterisk+1.8+Documentation , right?) |
| 17:08.23 | atheos_ | if you might be able to upload that to google docs if your PDF reader is broke |
| 17:08.42 | rutski | wait, yeah; found it |
| 17:10.34 | Katty | rutski: hi |
| 17:10.46 | rutski | hey |
| 17:10.49 | atheos_ | suzie_needs_help might be relevant https://issues.asterisk.org/view.php?id=12615 |
| 17:10.49 | Katty | what you been up to |
| 17:11.17 | Qwell | ~issue 1234 |
| 17:11.20 | Qwell | ~asteriskissue 1234 |
| 17:11.30 | Qwell | could've sworn there was some lookup he had |
| 17:12.07 | Katty | hi Qwell |
| 17:12.15 | Qwell | ohai |
| 17:12.18 | Katty | howrechu |
| 17:12.25 | WIMPy | M12615 |
| 17:12.35 | Qwell | Katty: still pretty upset about that thingie thing |
| 17:12.39 | Qwell | WIMPy: that bot was fixed |
| 17:12.40 | WIMPy | Hmm |
| 17:12.59 | WIMPy | Feared so :-( |
| 17:13.02 | Katty | Qwell: yeah :< |
| 17:13.06 | Qwell | russellb: we should put that bot in here |
| 17:13.10 | Qwell | maybe it is |
| 17:13.11 | Katty | Qwell: my thingie thing appears to be getting better |
| 17:13.13 | Qwell | ASTERISK-1234 |
| 17:13.14 | CaptainPants | [ASTERISK-1234] [Status: Closed] IAX taking down linked boxes - https://issues.asterisk.org/jira/browse/ASTERISK-1234 |
| 17:13.15 | Qwell | Katty: ! |
| 17:13.19 | Katty | Qwell: i know, right? |
| 17:13.21 | Qwell | russellb: NEVERMIND |
| 17:13.55 | irroot | ASTERISK-12AF <- Does he do HEX ?? |
| 17:13.56 | CaptainPants | [ASTERISK-12] [Status: Closed] app_voicemail2 became a bit silent, lately - https://issues.asterisk.org/jira/browse/ASTERISK-12 |
| 17:14.24 | irroot | ASTERISK-0x12AF <- Does he do HEX ?? |
| 17:14.27 | suzie_needs_help | what exactly does 'relaxdtmf=yes' do? |
| 17:14.33 | Qwell | suzie_needs_help: relaxes dtmf |
| 17:14.35 | WIMPy | Definitely pants :-) |
| 17:14.50 | Qwell | suzie_needs_help: it makes it so noisier dtmf can be detected |
| 17:14.59 | Katty | i like pants. |
| 17:15.02 | Qwell | only inband, obviously |
| 17:15.06 | Qwell | Katty: down with pants! |
| 17:15.07 | Katty | inband pants |
| 17:15.08 | suzie_needs_help | why wouldn't that just be built into DTMF ? |
| 17:15.08 | irroot | likes pants off :P |
| 17:15.24 | Qwell | suzie_needs_help: because it makes it so some female voices get detected as dtmf... |
| 17:15.24 | Katty | no i don't like pants off |
| 17:15.26 | Katty | unless it's hot |
| 17:15.37 | Katty | and then yay shower |
| 17:15.42 | Qwell | hawt |
| 17:15.56 | Katty | no, cold shower. |
| 17:15.57 | Katty | not hot. |
| 17:16.07 | suzie_needs_help | is it effective in dtmfmode=rfc2833 ? |
| 17:16.15 | Qwell | <Qwell> only inband, obviously |
| 17:16.31 | Katty | where's jbot |
| 17:16.50 | Katty | or, what are we calling jbot this year |
| 17:16.54 | Qwell | infobot: |
| 17:17.00 | Katty | infobot: inband |
| 17:17.23 | suzie_needs_help | does "DTMF begin ignored '2'" mean that 2 was ignored by asterisk? |
| 17:17.34 | Katty | infobot: inband is i like pants, inband pants. |
| 17:17.34 | infobot | Katty: okay |
| 17:18.13 | Katty | suzie_needs_help: sounds like it to me |
| 17:18.33 | Qwell | no, begins aren't really important, so it ignores them |
| 17:18.50 | Qwell | what you care about are the ends |
| 17:18.59 | Katty | the ENDS OF TIMES |
| 17:19.01 | Katty | i mean dtmfs |
| 17:19.26 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 17:19.26 | suzie_needs_help | so "DTMF end passthrough '2'" means the 2 was read by asterisk? |
| 17:20.09 | *** join/#asterisk joshaidan (~brianj@24.109.210.41) |
| 17:22.25 | *** part/#asterisk weinerk (~user@unaffiliated/weinerk) |
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| 17:28.34 | rutski | ok, so; say I want to find docs on a given item in the asterisk configs |
| 17:28.39 | rutski | like I see "pbx_realtime" in asterisk.conf |
| 17:28.44 | rutski | but I can't find it in the AST |
| 17:28.51 | rutski | are there other sources, from digium I mean |
| 17:29.12 | rutski | or did I just happen to stumble across an undocumented one? :-p |
| 17:30.01 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
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| 17:32.39 | LemensTS | how would you measure the ms of echo ? |
| 17:34.51 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 17:40.55 | wonko | oh nice, the AsteriskNOW install sets SELinux=0 |
| 17:40.59 | wonko | that's good |
| 17:41.13 | *** join/#asterisk fofware (~fabian@wdctf.siup.gov.ar) |
| 17:43.49 | Qwell | wonko: freepbx requires it |
| 17:44.02 | wonko | sane pepole require it |
| 17:44.05 | wonko | people |
| 17:44.27 | wonko | ok, FreePBX is pretty slick looking |
| 17:46.18 | *** join/#asterisk imox1234 (~imox1234@88.242.121.168) |
| 17:47.56 | leifmadsen | I dislike SElinux :) |
| 17:48.02 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 17:48.12 | leifmadsen | I understand what it does, but it always gets in my way and I've not taken the time to learn how to fix issues with it :) |
| 17:49.57 | wonko | selinux: making your computer impossible to use since 2003. |
| 17:51.57 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 17:56.00 | *** mode/#asterisk [+b *!*@gateway/web/freenode/ip.72.237.213.*] by Qwell |
| 17:56.07 | *** kick/#asterisk [suzie_needs_help!~north@pdpc/sponsor/digium/Qwell] by Qwell (suzie_needs_help) |
| 17:56.26 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 17:56.40 | jc319 | suzie needs unban joined #asterisk |
| 17:57.03 | Qwell | what? |
| 17:57.39 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 17:58.17 | Katty | brown chicken brown cow |
| 17:59.46 | n3hxs | How now? |
| 17:59.48 | Kobaz | i like brown cows |
| 17:59.57 | beek | I do too... cooked medium rare. |
| 18:01.30 | n3hxs | prefers rare. |
| 18:02.00 | Katty | moo |
| 18:04.26 | timholum1 | I have a strange issue, blind transfers to phone's does not work ( asterisk 1.8.3.3 ) if I dial 200 the phone rings, If I transfer to 200 the phone ring's I can answer it all is well, but if I blind transfer there is 30 seconds of dead air ( My timeout is 30 seconds ) then it goes to voicemail, But no ringing on extention 200 |
| 18:04.59 | timholum1 | I do get the line [Jun 21 12:59:24] WARNING[31620]: acl.c:698 ast_ouraddrfor: Cannot connect in the asterisk console |
| 18:05.02 | *** join/#asterisk Kevin24 (~kevin@190.99.204.24) |
| 18:05.09 | timholum1 | any idea's? |
| 18:06.04 | Qwell | my guess is it couldn't connect |
| 18:06.42 | timholum1 | I have done attempted this many times, and it is only on blind transfers |
| 18:06.55 | timholum1 | Does blind transfers connect using a different way? |
| 18:07.01 | Katty | did you try cookie offerings |
| 18:07.20 | timholum1 | or does it have a different context then reg transfer? |
| 18:09.25 | hardwire | can anybody find any problems with this: https://gist.github.com/6c020f5e5bc63098d02b |
| 18:09.49 | hardwire | the varchar(36) are uuids from external scripts |
| 18:10.02 | hardwire | I'm mostly concerned with partitioning. |
| 18:10.42 | hardwire | and combining the primary key with the start/eventtime |
| 18:10.49 | hardwire | which shouldn't really be a problem. |
| 18:12.29 | *** join/#asterisk Insonic (~kvirc@ip-178-203-122-117.unitymediagroup.de) |
| 18:13.18 | p3nguin | I don't know about "idea's" but I might have some ideas. |
| 18:16.33 | *** join/#asterisk jkroon (~jkroon@dsl-242-11-09.telkomadsl.co.za) |
| 18:17.01 | *** join/#asterisk jasoncarter (5006fcbf@gateway/web/freenode/ip.80.6.252.191) |
| 18:17.49 | jkroon | hi guys, how efficient is * at loading a few thousand extensions into a single context? |
| 18:18.19 | jasoncarter | Is it possible to connect a standard telephone and fax machine through a 56k modem port of my computer and connect it to a remote asterisk server? |
| 18:18.25 | jkroon | i'd like to set up a context that sets up descriptions based on country, area and other prefix codes and the dialplan mechanism seems about the sanest way to do this but it seems to take forever to load. |
| 18:18.49 | _Corey_ | jkroon: Take a look at Asterisk realtime |
| 18:18.50 | hardwire | jkroon: if * dialplan isn't working out well.. you can always use AGI |
| 18:18.52 | jkroon | jasoncarter, if it is possible I do not know how. I seriously doubt it. |
| 18:18.53 | hardwire | or realtime |
| 18:19.19 | hardwire | or a predictable extension pattern |
| 18:19.32 | hardwire | or macros with lookups |
| 18:19.42 | jkroon | hardwire, predictable extension pattern? |
| 18:19.52 | hardwire | _1907NXXXXXX |
| 18:20.06 | hardwire | thats more than a few thousand :) |
| 18:20.34 | jkroon | hardwire, my patterns will be things like _27! and _2712! and _2712348! type of things :) |
| 18:20.35 | hardwire | like.. if you're planning a product.. and all you need to do is set an accountcode or something simple.. it can be reduced dramatically. |
| 18:20.38 | jkroon | so no :p |
| 18:20.43 | Kevin24 | jasoncarter. some PCI 56k modems can be used as FXS on asterisk |
| 18:20.45 | hardwire | jkroon: checked out a2billing? |
| 18:20.58 | jkroon | hardwire, yes and ditched it. |
| 18:21.06 | _Corey_ | jkroon: What's your objective with the dial plan? |
| 18:21.11 | jkroon | billing engine is inadequate. |
| 18:21.37 | jasoncarter | kevin24 thank you kindly |
| 18:21.42 | hardwire | jkroon: then you should probably test out loading that many prefixes into a dialplan to see for yourself. |
| 18:21.43 | jkroon | _Corey_, just want to set call descriptions, so for the above _27 would just set a description of ZA, the 2712 would set Pretoria and so forth. |
| 18:21.45 | hardwire | :) |
| 18:22.13 | jkroon | hardwire, last time I tried to load around 30k prefixes it took about 20 odd minutes. |
| 18:22.18 | jkroon | so would prefer to avoid that :) |
| 18:22.20 | hardwire | jkroon: I have a solution for that actually |
| 18:22.28 | _Corey_ | jkroon: I'd find the dial plan approach a little inflexbile; I'd suggest an AGI w/a SQL lookup that can deal with patterns and "best match" better |
| 18:22.33 | jkroon | please do tell ? |
| 18:22.43 | Kevin24 | jasoncarter, check out http://forums.whirlpool.net.au/archive/841396 |
| 18:22.50 | jkroon | was looking at astdb but it can't do longest prefix. |
| 18:22.56 | hardwire | I pregenerated verbose names for zones and stored them as /2/7/1/2/3/4/8 on the filesystem |
| 18:23.02 | hardwire | it made for really fucking fast lookups |
| 18:23.15 | hardwire | I just wrapped a macro around it to find the longest existing path. |
| 18:23.23 | jkroon | that is very sneaky! |
| 18:23.28 | hardwire | wrote a quick python script to break it out |
| 18:23.42 | hardwire | what numbering plan are you using? |
| 18:23.46 | hardwire | if you don't mind me asking |
| 18:24.24 | jkroon | e164, gotta deal with the "global" situation :p |
| 18:24.27 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 18:24.33 | hardwire | err.. what numbering plan provider |
| 18:24.38 | hardwire | the database provider |
| 18:24.47 | hardwire | I'm looking for alternatives |
| 18:24.50 | jkroon | hehe, scraping it off of various locations. |
| 18:24.57 | hardwire | ah ok |
| 18:25.12 | hardwire | talk to you next year then :) |
| 18:25.13 | jkroon | was actually communicating with a guy in the UK to help so that we can start building up an "open" list |
| 18:25.24 | hardwire | I own opendial.org |
| 18:25.26 | hardwire | it was supposed to do that |
| 18:25.33 | hardwire | let me know if you wanna work on it and I'll hand it over. |
| 18:25.58 | jkroon | I can probably get hold of the ZA list from ICASA, he has a reasonable list for the UK from the sound of it. |
| 18:26.00 | Kevin24 | i need help on how to get a VM * box to another * box trunk using SS7 for a test platform. can it be done with software only? |
| 18:26.17 | hardwire | Kevin24: rsync |
| 18:26.22 | hardwire | and MWI file scanning |
| 18:26.29 | hardwire | heh |
| 18:26.44 | *** join/#asterisk vfabi (~fabi@host-static-188-237-247-198.moldtelecom.md) |
| 18:26.49 | Kevin24 | hardwire, thanks |
| 18:26.50 | hardwire | not sure how to use SS7 to do that without voice degregation unless you are using a soundmodem. |
| 18:26.56 | jkroon | hardwire, so you reckon it's unfeasible to build such a list? |
| 18:27.04 | hardwire | jkroon: it's highly feasable |
| 18:27.11 | hardwire | given enough time |
| 18:27.11 | jkroon | but ... ? |
| 18:27.18 | hardwire | the list I have of breakouts is well ove 3 million |
| 18:27.23 | jkroon | and enough participation obviously. |
| 18:27.28 | hardwire | there are larger lists out there |
| 18:27.37 | hardwire | they change constantly |
| 18:27.41 | hardwire | esp during war. |
| 18:27.56 | jkroon | that's problematic. |
| 18:28.17 | jkroon | and the NANP doesn't make things easier either. |
| 18:28.31 | hardwire | well you's use the ITU |
| 18:28.34 | hardwire | you'd |
| 18:28.44 | hardwire | for a good base of it.. |
| 18:28.58 | hardwire | I'd much rather have a free database |
| 18:29.00 | hardwire | much rather |
| 18:29.52 | hardwire | Kevin24: did that makes sense.. I'm not sure how familiar peeps are with the voicemail app.. but it can scan spool dirs if you configure it to. |
| 18:29.58 | jkroon | well, question - can people in general be trusted to make sane additions? i mean, scrubbing wikipedia for at least the country codes is relatively simple, getting a list from ICASA for initial ZA population is simple enough for me, but that is a VERY small start. |
| 18:30.13 | hardwire | jkroon: karma |
| 18:30.21 | hardwire | and unique identifiers |
| 18:30.22 | hardwire | and rollbacks |
| 18:30.32 | *** join/#asterisk scalex000 (~chatzilla@190.166.84.26) |
| 18:30.34 | Kevin24 | hardwire, it give me someplace to start, thanks |
| 18:30.54 | jkroon | ok, i'll sleep on it. you hang here often in case I'd like to contact you again? |
| 18:30.57 | hardwire | if somebodies karma appears low then you can opt to roll back their commits to the db until they can become accredited again through other users. |
| 18:31.35 | hardwire | Kevin24: it's exciting :) You can always use dialplan to another location and DTMF entry to trigger the events required to route the voicemail. |
| 18:32.03 | hardwire | like.. hey other system.. I have a voicemail for you.. it's uuid is 0013482459345993495934599399345 |
| 18:32.18 | hardwire | and it is expected to look at http://beepboopbeep.com/0013482459345993495934599399345 |
| 18:32.28 | jkroon | rofl @ hardwire sounds like a messed up scheme I use to figure out analog lines's numbers :p |
| 18:32.35 | hardwire | jkroon: damn straight |
| 18:32.43 | Kevin24 | lol |
| 18:32.43 | jkroon | dial in to a known number and request it to play back the detected CLI using DTMF ;p |
| 18:33.12 | jkroon | still working on a scheme to figure out the exchange prefix. |
| 18:33.19 | Kevin24 | could i use this to call a list of numbers and telemarket? Like Homer did? |
| 18:33.38 | *** join/#asterisk fofware (~fabian@wdctf.siup.gov.ar) |
| 18:33.39 | jkroon | suspect the simplest may be to detect the CLI then dial it out on the ISDN and wait for the return call, see how many of the tail digits you receive. |
| 18:33.39 | *** join/#asterisk d-_-b- (~d-_-b-@2607:f370:9999:dead:5ab0:35ff:fef7:6be3) |
| 18:33.59 | jkroon | use some obscure (non-speech) transfer capability to ID the call. |
| 18:35.26 | jkroon | hardwire, ok, how would you like to see the DB exposed? Using an ENUM like scheme using TXT RRs, HTTP GET (restful) or some other API? Downloadable list? |
| 18:35.31 | hardwire | sound modems are the bomb btw. |
| 18:35.33 | jkroon | subscriptions for updates? |
| 18:36.01 | hardwire | jkroon: lots of ways. decentralized databases, dns, paid gateways, etc.. |
| 18:36.10 | hardwire | bittorrent :) |
| 18:36.28 | jkroon | hardwire, primary aim would be to make it available free of charge, and yea - i like the BT idea for distribution. |
| 18:37.11 | jkroon | ok, i'll sleep on it. back to getting faxing working. |
| 18:39.25 | hardwire | so y'all think my cdr/cel tables are saneish? |
| 18:39.45 | hardwire | I'm kinda worried about using timestamp, but I need to be able to set the session timezone without using convert_tz all over the place |
| 18:40.06 | hardwire | checks mysql logs to see if cdr_odbc and cel_odbc sets the session timezone before submitting inserts |
| 18:40.26 | jkroon | ok, back to realtime (used here for another purpose of remapping prefixes, I'll receive anumber like 0123456789 and I need to replace the 012345 prefix with eg, 086321), is it possible to use multiple tables for different contexts? The examples I have use a single table for all contexts ? |
| 18:40.57 | hardwire | GOTO(CONTEXT-${EXTEN:2},${EXTEN},1) |
| 18:41.40 | hardwire | but in reality.. all you'd need would be a small set of dialplan to remap the numbers based on stored data. |
| 18:43.13 | jkroon | hmm, ok, so the remaps is probably a bad example, but the point being I've got one table with a list of "extensions" and another with a bunch of remaps, and from the looks of it I can't easily use multiple tables for two different contexts ..., so remaps would be [remap] switch => Realtime/remap or something and extensions [extensions] switch => Realtime/exts ... but using different tables in mysql somehow |
| 18:45.12 | hardwire | let func_odbc be your guide. |
| 18:45.26 | hardwire | if you don't want to use realtime |
| 18:45.42 | hardwire | or AGI/FastAGI |
| 18:45.53 | hardwire | or a dir based table. |
| 18:46.49 | jkroon | i'm going to need realtime for the SIP peers anyway, so can just as well use realtime for this too. the switch => syntax is useful because the VALID_EXTEN function should still work properly. |
| 18:52.10 | jkroon | ok, look like just using func_odbc might turn out to be simpler, assume the data is there, look it up, and if it fails go to alternate mechanism. |
| 18:53.19 | *** part/#asterisk LemensTS (~matthew@adsl-70-238-135-100.dsl.stlsmo.sbcglobal.net) |
| 18:54.45 | *** join/#asterisk mha_ (~mha@109.161.140.230) |
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| 19:07.16 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 19:09.20 | Katty | has cookies and chips! and is still losing weight! |
| 19:09.42 | Katty | tho i haven't had any for probably a month |
| 19:09.44 | Katty | but still |
| 19:09.47 | jaytee | has cheese and is getting visibly larger with each passing moment |
| 19:10.01 | Katty | jaytee: mmm cheese |
| 19:10.30 | Kevin24 | sounds like a infection? "has cheese and is getting visibly larger" |
| 19:10.42 | Katty | dear liver. |
| 19:10.53 | Katty | i appologize in advance for the 16g of sugar i am about to stuff down you |
| 19:10.57 | Katty | it was for a good cause |
| 19:10.59 | Katty | love, katty |
| 19:13.04 | *** join/#asterisk celord (~celord@200.91.75.115) |
| 19:13.16 | Katty | these cookies are amazing |
| 19:14.13 | Qwell | Katty: where are mine? :( |
| 19:16.26 | mha_ | Installed asteriskNOW, but don't know the user/password for web page? |
| 19:17.36 | russellb | have you asked the google? |
| 19:17.48 | Qwell | Have you read the getting started guide, that was linked from the download page? |
| 19:18.24 | *** join/#asterisk imox1234 (~imox1234@88.242.121.168) |
| 19:18.27 | carrar | PICS of said cookies |
| 19:18.34 | mha_ | russ: cannot find a user/pass in digium either |
| 19:19.26 | Katty | Qwell: come get them |
| 19:19.26 | russellb | http://www.asterisk.org/AsteriskNOW-1.5-QuickStart |
| 19:19.36 | Qwell | Katty: don't tempt me |
| 19:19.44 | Qwell | don't think I won't. :P |
| 19:19.59 | beek | Have any coffee to dunk those cookies in? |
| 19:20.01 | Qwell | russellb: I need Wednesday off so I can go stalk Katty. |
| 19:20.16 | Katty | :< |
| 19:20.22 | Qwell | stalk in a good way! |
| 19:20.34 | russellb | Qwell: i'm not your manager anymore, heh |
| 19:20.40 | Qwell | quiet you |
| 19:20.49 | russellb | but sure, go for it |
| 19:23.23 | Katty | carrar: imagebin.org/159381 |
| 19:23.55 | Qwell | that's some serious shutter latency |
| 19:24.07 | cj | so. anyone want to help me figure out how to work DUNDi in to my system? |
| 19:24.11 | Qwell | IRC scrolled and ghosted in the background |
| 19:24.12 | Katty | it was my phone. |
| 19:24.31 | cj | I've got two * machines in a load balanced pool |
| 19:24.52 | cj | incoming REGISTER requests get round-robin'd across them |
| 19:25.16 | Katty | i wonder where the term round robin came from |
| 19:25.21 | cj | say user/extension 4016 gets sent to asterisk0 and 4012 goes to asterisk1 |
| 19:25.33 | cj | Katty: *shrug* :) |
| 19:25.43 | Qwell | Katty: document signing |
| 19:26.06 | cj | I've got asterisk1 and asterisk0 configured using DUNDi |
| 19:26.15 | Katty | Qwell: eh? |
| 19:26.42 | Qwell | Katty: round ribbon, a method of signing a document by a group of people, so as not to give away the order in which it was signed. |
| 19:26.49 | Katty | ohhh |
| 19:26.59 | Katty | so how did it go from ribbon to robin |
| 19:27.04 | Qwell | 'mericans |
| 19:27.15 | Katty | oh so we botched that one up? |
| 19:27.18 | Katty | awesome. |
| 19:27.29 | Qwell | it was french or something |
| 19:27.33 | cj | http://paste2.org/p/1482467 |
| 19:27.52 | _Corey_ | comes here for the trivia |
| 19:29.18 | cj | dundi lookup seems to work fine, too: |
| 19:29.18 | cj | http://paste2.org/p/1482468 |
| 19:29.55 | n3hxs | ribbon - robin just a typo on their old machines. |
| 19:31.27 | cj | as phones register, they get dumped in the RegisteredDevices context |
| 19:31.58 | cj | and dundi is configured to look up extensions from the RegisteredDevices context: |
| 19:32.01 | cj | extensions => RegisteredDevices,0,IAX2,dundi:SECRET@asterisk0.sipnet.ntr/${NUMBER},nopartial |
| 19:32.16 | cj | extensions => RegisteredDevices,0,IAX2,dundi:SECRET@asterisk1.sipnet.ntr/${NUMBER},nopartial |
| 19:32.44 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 19:34.12 | cj | also, I have extensions in the RegisteredDevices context with NoOp() entries for all phones on both peers: |
| 19:34.53 | cj | exten => 4012,1,NoOp() |
| 19:34.53 | cj | exten => 4016,1,NoOp() |
| 19:35.20 | cj | once a user registers, asterisk tells me that this changes from NoOp() to Noop(<extension) |
| 19:35.23 | carrar | mm cookies |
| 19:35.54 | Katty | carrar: they were very mmm |
| 19:35.57 | Katty | carrar: and now very gone |
| 19:36.03 | carrar | woah! |
| 19:36.08 | cj | hi carrar! |
| 19:36.17 | carrar | hi cj! |
| 19:36.32 | carrar | Pouring down rain in Seattle, typical weather! |
| 19:36.34 | cj | want to come to orcas next tuesday morning for our clec meeting? :) |
| 19:36.41 | cj | wut. It's beautiful in Eastsound :) |
| 19:36.45 | carrar | heh |
| 19:36.45 | Katty | i like orcas |
| 19:36.52 | carrar | too hot here too |
| 19:37.04 | Katty | what do orcas have to do with meetings |
| 19:37.06 | carrar | clec meetings!?? |
| 19:37.09 | Katty | meeting of the orcas? |
| 19:37.18 | carrar | orca orca! |
| 19:37.18 | Katty | would be down for orca meeting |
| 19:37.33 | carrar | You have to dress like a native indian Katty |
| 19:37.49 | Katty | you mean a native orca |
| 19:37.57 | Katty | i have some black cream eyeshadow that might do the trick |
| 19:38.02 | cj | carrar: 2nd & 4th Tuesday mornings after the WA Dept. Emergency Mgmt. radio equipment at station 21, across the street from the airport |
| 19:38.04 | Katty | and plenty of black eyeliner |
| 19:38.05 | carrar | heh |
| 19:38.27 | cj | the word test should have made it in there somewhere. probably after 'equipment' |
| 19:38.43 | carrar | It's not in Bellevue? |
| 19:39.10 | carrar | Sounds too far! |
| 19:39.18 | Katty | according to google, a healthy young orca costs about a million dollars |
| 19:39.20 | cj | no, but there are flights from the northeast end of lake wa. to the airport across the street :) |
| 19:39.32 | Katty | or you can trade 2 belugas |
| 19:39.34 | cj | $120 one way or something |
| 19:39.36 | Katty | for 1 orca |
| 19:40.21 | cj | maybe we'll set up a video conference call :) |
| 19:40.31 | carrar | ok time to go walking for a hour with my 35lb pack |
| 19:40.38 | cj | carrar: enjoy! |
| 19:40.39 | carrar | video call is good |
| 19:41.21 | cj | so, back to asterisk... anybody here ever used DUNDi to allow two * machines to share their extensions? |
| 19:43.03 | *** join/#asterisk flyman_ (~flyman_@194.24.158.1) |
| 19:45.08 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 19:46.16 | russellb | yup |
| 19:46.24 | russellb | there is a DUNDi chapter in: |
| 19:46.25 | russellb | ~thebook |
| 19:46.25 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 19:47.44 | cj | russellb: yeah, I've tried it... it doesn't work. :( |
| 19:47.57 | cj | russellb: I think there's a problem with the SIP dundi transport in 1.8 |
| 19:48.06 | cj | using IAX2 seems to work |
| 19:48.15 | russellb | DUNDi doesn't transport SIP or IAX2 |
| 19:48.19 | russellb | it transports a URI |
| 19:48.39 | cj | erhm... it sends that URI over.... |
| 19:48.39 | russellb | so it would just be a configuration thing |
| 19:48.49 | russellb | that the URI that is configured is not a usable one |
| 19:49.08 | cj | how does it communicate that URI from asterisk0 to asterisk1? |
| 19:49.14 | russellb | using DUNDi :-) |
| 19:49.22 | cj | ok, so what's the SIP in this mean? |
| 19:49.36 | cj | extensions => RegisteredDevices,0,IAX2,dundi:SECRET@asterisk1.sipnet.ntr/${NUMBER},nopartial |
| 19:49.37 | russellb | the URI come from the mappings configuration in dundi.conf |
| 19:49.40 | cj | ;extensions => RegisteredDevices,0,SIP,dundi:SECRET@asterisk1.sipnet.ntr/${NUMBER},nopartial |
| 19:50.05 | russellb | that's not a valid SIP dial string |
| 19:50.15 | cj | ok. what's a valid one look like? |
| 19:50.17 | russellb | take a step back and put what you configure into Dial() directly |
| 19:50.26 | russellb | and once you get that working, then put that in dundi.conf |
| 19:50.34 | russellb | the syntax is in sip.conf.sample |
| 19:51.01 | russellb | for your example there, it would be something like .... |
| 19:51.06 | cj | exten => 1000,1,Verbose(2,Test extension to place call to asterisk0) same => n,Dial(IAX2/dundi:SECRET@asterisk0.sipnet.ntr/1000,30) same => n,Hangup() |
| 19:51.12 | cj | irssi-- |
| 19:51.55 | russellb | ...SIP,${NUMBER}:SECRET::dundi@asterisk1.sipnet.ntr,nopartial |
| 19:52.14 | russellb | ; SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port] |
| 19:52.20 | russellb | the syntax isn't quite the same as IAX2 |
| 19:52.33 | cj | ok. I'll try switching back to SIP |
| 19:52.51 | russellb | and make sure you have a peer called [dundi] configured in sip.conf |
| 19:54.34 | cj | http://paste2.org/p/1482501 |
| 19:55.42 | Katty | weeeeeeeeeeeee |
| 19:55.44 | Katty | boingboingboing |
| 19:55.50 | Katty | WARNING HIGHLY CAFFEINATED |
| 19:55.54 | Katty | dances with russellb |
| 19:56.30 | jplank | seems like someone is bored |
| 19:57.37 | jaytee | wonders if Network Solutions will ever get their shit together |
| 19:57.41 | cj | http://paste2.org/p/1482513 |
| 19:57.57 | beek | that's a rhetorical question, right jaytee? |
| 19:58.03 | jaytee | beek, yes |
| 19:58.07 | cj | jaytee: I don't think I know anybody at Network Solutions at this moment, so I can't tell you. |
| 19:58.28 | cj | they've got offices in Bellingham, though, I think. |
| 19:58.50 | cj | I think I'll be CLECing it up in their neck of the woods in the near future |
| 19:58.59 | jaytee | had to go to GoDaddy to get an SSL cert for a website. The original was with NS but after spending most of the day yesterday listening to excuses about their system being offline I got tired of waiting. |
| 19:59.12 | cj | maybe they'll fly over and see me. $45 flight one way. |
| 19:59.19 | cj | jaytee: no. |
| 19:59.26 | Katty | $45 flight? meep |
| 19:59.31 | jplank | godaddy is worse then netsol |
| 19:59.31 | Katty | think i'll stay on the ground |
| 19:59.36 | *** join/#asterisk jeffspeff (~jeffspeff@173.11.144.149) |
| 19:59.44 | cj | jaytee: if startssl didn't suspend their issuance recently, I'd tell you to go talk to them. |
| 19:59.53 | jeffspeff | i'm looking into using the new office 365 when that's released, but was wanting to know if its "enterprise voice capabilities" could work with my Asterisk SIP server |
| 20:00.04 | cj | but they're being audited by the military for issuing me a cert without checking my ID or something ;) |
| 20:00.10 | jaytee | jplank, at this point I'm beginning to think that "They all suck!" fits in more ways than one. |
| 20:00.34 | jplank | your probably right about that, but I've had great experiences with enom. |
| 20:01.11 | jplank | well, let me rephrase that, I haven't had any long term outages, and got responses from the customer service when I had an issue transferring a domain to them. |
| 20:02.17 | mboylan | quick question guys - if I have phone1 sip registered to asteriska which is connected to asteriskb via IAX2 calling phone2 sip registered to asteriskb, is it possible for the asterisk/iax bridge to fall out of the middle and allow the point to point SIP connection? |
| 20:03.00 | mboylan | both phone1 and phone2 have canreinvite=yes in their sip entries on asteriska and asteriskb |
| 20:03.06 | jplank | are they behind NAT? |
| 20:03.09 | mboylan | no |
| 20:03.21 | jplank | then you should be able to reinvite asterisk out of the way |
| 20:03.32 | p3nguin | Doesn't matter. Asterisk will still remain in control of the signaling (SIP). |
| 20:03.43 | jplank | I think he's talking about RTP |
| 20:04.14 | jplank | if not, then just use URI dialing |
| 20:04.18 | mboylan | So even if the RTP stream is point to point, if asterisk is shutdown in the middle, the call will drop? |
| 20:04.45 | p3nguin | potentially |
| 20:04.50 | jplank | well, no |
| 20:05.02 | jplank | if you reinvite asterisk out of the media path |
| 20:05.09 | p3nguin | As soon as the phones try to get asterisk to do something on their behalf, that will fail. |
| 20:05.10 | jplank | and asterisk goes down, the media will keep flowing |
| 20:05.26 | jplank | exactly, as soon as it tries to do another SIP message it will fail |
| 20:05.51 | mboylan | If you weren't in here earlier... we're upgrading campus from 1.2 BE to 1.8 and have a distributed environment. We want to redo the routing layer first, but we're finding some weird incompatibilities with 1.2 and 1.8 in terms of SIP trunking. The reinvites don't seem to be handled properly down the trunk |
| 20:06.13 | jplank | there are huge differences between 1.2 to 1.8 |
| 20:06.33 | jplank | I think reinviting in 1.8 is different |
| 20:06.37 | mboylan | so after we've spent about a week ruling out networking as a cause, boss finally said maybe we'll still have to use IAX until we can get everything up to 1.8 |
| 20:06.59 | jplank | I only have one server using 1.8, but isn't it NOT canreinvite=yes/no anymore? |
| 20:07.17 | mboylan | jplank: It still works, but it's directmedia now afaik |
| 20:07.49 | jplank | something like that, but I think it works differently now. I haven't tried it on 1.8, but I could of swore from the release notes it works differently |
| 20:08.49 | mboylan | jplank: There's lots of options. There's a new 'directrtpsetup' option now too |
| 20:08.59 | mboylan | from what I can tell, our 1.8 to 1.8 trunks are working fine |
| 20:09.02 | mboylan | calls aren't dropping |
| 20:09.55 | mboylan | but from the 1.2 to 1.8, you get some sort of constant reinvite but the ids never change. It looks like the same invite over and over so it can never properly be acknowledged |
| 20:10.08 | *** join/#asterisk jaybee_ (~jortbloem@ip-210-48-105-162.iconz.net.nz) |
| 20:10.21 | jplank | do you havea capture I could look at? |
| 20:11.07 | jaybee_ | hi all. Is there a table of asterisk exit codes somewhere? |
| 20:11.25 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 20:12.31 | wdoekes2 | application exit codes? I don't think there are that many for asterisk |
| 20:13.15 | jaybee_ | never mind, it's not asterisk exit code... I'll keep looking |
| 20:13.57 | Kobaz | what's a quick number i can try for international calling |
| 20:14.05 | Kobaz | uk or somethig |
| 20:14.14 | jplank | do you mean hangup cause codes? ISDN cause codes? |
| 20:14.43 | jaybee_ | I meant number returned to unix when the process terminates |
| 20:14.50 | mboylan | jplank: Yeah, I do actually |
| 20:14.56 | jplank | ahhh |
| 20:15.13 | _Corey_ | Kobaz: I usually look up a hotel... international chains are the best |
| 20:15.20 | jplank | mboylan: I wouldn't mind taking a look unless your already know for a fact why its dropping |
| 20:15.25 | mboylan | jplank: This is on the 1.8 side of things -- http://pastie.org/private/csfcenzp4xwkzznib192ha |
| 20:15.28 | mboylan | no, please do |
| 20:17.53 | *** join/#asterisk AviMarcus (~avi@bzq-79-182-184-57.red.bezeqint.net) |
| 20:18.01 | mboylan | 10.1.1.15 is the 1.2 asterisk box. 10.96.192.14 is the new 1.8 box |
| 20:18.03 | jplank | just to make sure I'm reading this right, 10.1.1.15 is the far end asterisk? |
| 20:18.15 | jplank | aha ok |
| 20:18.15 | AviMarcus | are there any UK mobile test numbers? I just added a termination carrier, I want to test my routes.. |
| 20:19.05 | jaybee_ | AviMarcus, just do what we do: google "pizza <location>" |
| 20:19.33 | mboylan | jplank: On the 1.2 console errors would show about SIP 491 Request Pending or something like that |
| 20:19.35 | Kobaz | _Corey_: mm, k |
| 20:19.46 | mboylan | and then finally in the 1.8 console you see the "No reply to our critical packet" |
| 20:19.48 | jaybee_ | 0844 371 2550: Fire & Stone Pizza, Covent gardens |
| 20:19.59 | Katty | mmmm pizza |
| 20:20.08 | jaybee_ | exactly! |
| 20:20.10 | mboylan | we turned off IPTables on both and the new box currently is on a completely open subnet. As are the phones. Shouldn't be a networking issue |
| 20:20.18 | AviMarcus | 0844 isn't mobile, that's a special route |
| 20:20.32 | AviMarcus | 0447[1-9] |
| 20:20.59 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 20:21.04 | wdoekes2 | mboylan: looks like the 1.2 asterisk doesn't ACK the 491 and the 1.8 considers the call dead |
| 20:21.05 | AviMarcus | thx though I'll try thaat |
| 20:21.34 | mboylan | wdoekes2: Would you say this is probably just an unfortunate incompatibility? |
| 20:22.16 | wdoekes2 | I've seen similar things break with 3 1.8's in a row.. but there the RTP stream went south |
| 20:22.32 | wdoekes2 | can you canreinvite=no on the 1.2? |
| 20:22.56 | mboylan | I guess I could on my phone. 1.2 is production so I have to be careful |
| 20:23.07 | mboylan | If I do it on the 1.8 side, asterisk stays in the middle, and the call stays up |
| 20:25.00 | p3nguin | drmessano: Just a follow-up on Google Voice call screening that we briefly discussed a couple weeks ago... I've tried switching screening on and off (for my forwarding numbers), and there is no change at all. If it's set to on, it sometimes screens and sometimes does not; the same is true with screening set to off. |
| 20:25.05 | mboylan | wdoekes2: It looks like 1.2 ACKs the 491 at least at some point in there, no? |
| 20:25.39 | mboylan | after the first one |
| 20:25.45 | mboylan | then the subsequent ones fail |
| 20:25.53 | wdoekes2 | no.. you should see CSeq: 102 ACK received |
| 20:26.26 | mboylan | ah, ok |
| 20:26.32 | *** join/#asterisk radic (~radic@dslb-178-002-218-201.pools.arcor-ip.net) |
| 20:27.11 | mboylan | know of any workarounds or is it off to IAX until everything gets to 1.8? |
| 20:27.29 | wdoekes2 | I'm afraid to say IAX |
| 20:27.50 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 20:28.09 | *** part/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 20:28.15 | wdoekes2 | I don't think you'll get both asterisken out of the RTP |
| 20:28.36 | mboylan | wdoekes2: From the testing I did earlier today it seems that's the case, yeah |
| 20:29.01 | mboylan | But at least it'd only be temporary until we get the other boxes up to 1.8 as well |
| 20:32.59 | *** join/#asterisk radic (~radic@dslb-178-002-218-201.pools.arcor-ip.net) |
| 20:33.02 | hardwire | so with cdr_custom.conf if I wanted to ensure gmtime for times.. I would simply make a string out of the timestamp and convert it back to a timestamp with a specific timezone reflecting the local timezone. I think. |
| 20:33.06 | hardwire | tests |
| 20:33.22 | hardwire | can ignore this.. since he uses UTC on every system... but still. |
| 20:36.07 | *** join/#asterisk manji (~manjiki@2a02:580:8000:d001:226:bbff:fe13:1c09) |
| 20:36.20 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:37.37 | *** part/#asterisk jaybee_ (~jortbloem@ip-210-48-105-162.iconz.net.nz) |
| 20:39.04 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 20:44.02 | *** join/#asterisk jeroe (~jr@p57AC7234.dip.t-dialin.net) |
| 20:45.31 | *** part/#asterisk jeroe (~jr@p57AC7234.dip.t-dialin.net) |
| 20:52.21 | *** part/#asterisk AviMarcus (~avi@bzq-79-182-184-57.red.bezeqint.net) |
| 20:55.44 | *** join/#asterisk celord (~celord@201.198.102.2) |
| 20:55.44 | *** join/#asterisk MiserySoft (~MiserySof@94-116-26-46.dynamic.thecloud.net) |
| 20:57.03 | DrDigital | Wiretap7 |
| 20:57.46 | WiretapWork | hi |
| 21:02.51 | *** join/#asterisk breardo (~breardo@234-200-29-134.hcc.mnscu.edu) |
| 21:02.59 | breardo | hello.. |
| 21:03.00 | hardwire | lol |
| 21:03.07 | hardwire | http://i.imgur.com/VXyUA.jpg |
| 21:03.11 | hardwire | cracks himself up |
| 21:03.14 | breardo | having an issue getting 1.8.4.2 to compile with LibPRI 1.4.11.5 (but 1.4.12-beta5 works!) |
| 21:03.23 | breardo | here is my ./configure error output: http://pastebin.com/hVfNaHBu |
| 21:03.38 | breardo | here is my config.log from the failed build: http://pastebin.com/hVfNaHBu |
| 21:03.41 | breardo | doh |
| 21:03.49 | breardo | http://pastebin.com/yfuZ7wR9 |
| 21:05.40 | breardo | anyone know if it's "safe" to use libPRI 1.4.12-beta5? Will that release become 1.4.12? |
| 21:05.45 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 21:08.29 | *** part/#asterisk Kevin24 (~kevin@190.99.204.24) |
| 21:09.58 | sawgood | Can anone offer a tip or two on how I can get some additonal columns into /var/log/asterisk/cdr-csv/Master.csv (for example the Asterisk hangupcause code)? |
| 21:10.23 | breardo | sorry I mean 1.4.12-beta3 |
| 21:10.39 | Qwell | sawgood: use cdr_custom instead |
| 21:10.54 | sawgood | Hi Qwell ... thank you ... |
| 21:11.10 | sawgood | are the columns for cdr-custom controlled from /etc/asterisk/cdr.conf? |
| 21:23.25 | *** join/#asterisk voxio (~mark@unaffiliated/voxio) |
| 21:23.33 | *** join/#asterisk flyman_ (~flyman_@194.24.158.1) |
| 21:24.20 | voxio | hey guys.. anyone running asterisk in EC2? I've set-up a large instance with asterisk 1.8 using a 1000hz kernel supplied by amazon but I'm still getting some choppy call issues. |
| 21:25.15 | voxio | I'm using the standard "amazon linux" AMI |
| 21:25.22 | voxio | and the latest 1.8 |
| 21:25.33 | WiretapWork | voxio, are you tagging the RTP properly? |
| 21:26.03 | voxio | WiretapWork: probably not .. would that be a firewall setting in EC2? |
| 21:26.35 | WiretapWork | voxio, you will need to tag your RTP traffic with some DSCP tag most probably to ensure you get proper prioritisation |
| 21:26.39 | wwalker | voxio: I've run stuff there before. it was OK, but not great. I've not used 1.8, but I'd guess that an increased jitter buffer and a nice -n -10 asterisk might help |
| 21:26.44 | WiretapWork | there is no guarantee your ISP will respect that, but eh |
| 21:27.09 | WiretapWork | one of the problems we have here is our ISP untags inbound packets |
| 21:27.19 | WiretapWork | so inbound audio is choppy as hell, outbound is crystal clear |
| 21:27.51 | voxio | I thought the problem would be down to it being a virtualized (xen) enviroment |
| 21:28.19 | voxio | as I've heard there are problems with virtualized environments and asterisk due to timing |
| 21:28.20 | WiretapWork | possibly, but also possibly not |
| 21:28.35 | WiretapWork | there can be, yes, I've heard xen particularly |
| 21:28.41 | WiretapWork | personally VMWare has worked perfectly for me |
| 21:28.57 | voxio | are there any kernel modules that help with timing? |
| 21:29.36 | voxio | wwalker, I'll try that out |
| 21:29.52 | voxio | I'm only testing at the moment as we have two sites, one in california and one in the UK |
| 21:30.08 | voxio | they both ping around 70MS to the east coast (virginia) |
| 21:30.42 | voxio | so I was thinking of consolidating into one server and having a hot backup in another EC2 datacentre with IAX failover |
| 21:31.26 | voxio | the connection between the two sites can be poor at times (UK to west coast) |
| 21:31.55 | voxio | and the US users are currently connecting over to the UK server |
| 21:32.06 | voxio | via SIP |
| 21:37.31 | *** join/#asterisk Kevin24 (~kevin@190.99.204.24) |
| 21:37.56 | Kevin24 | Hi Folks, How can i get CID to pass though to SS7,, using libss7 |
| 21:38.06 | *** join/#asterisk quintana (~sylvain@aghnar.doowan.net) |
| 21:38.36 | WIMPy | Kevin24: It might be hard to avoid, I guess. |
| 21:39.21 | Kevin24 | the ss7 link is showing the CID of the E1 not the calling party |
| 21:40.15 | WIMPy | Which E1? |
| 21:40.39 | WIMPy | I think we need more information. |
| 21:40.48 | Kevin24 | does this take effect on the libss7 usecallerid=yes callerid=asrecieved .. or does libss7 override this? |
| 21:41.20 | Kevin24 | on the T1/E1 interface. |
| 21:41.27 | Kevin24 | drawing map |
| 21:42.17 | WIMPy | _guesses_ your using an E1 via dahdi as a clear channel with SS7 on top. |
| 21:42.49 | *** join/#asterisk d-_-b- (~d-_-b-@2607:f370:9999:dead:5ab0:35ff:fef7:6be3) |
| 21:43.47 | Kevin24 | yes |
| 21:43.52 | *** part/#asterisk wesphillips (~wphill04@137.237.233.124) |
| 21:44.19 | *** join/#asterisk REdOG (~REdOG@gentoo/user/redog) |
| 21:45.10 | WIMPy | I don't think any dahdi configuration is used above the interface layer. |
| 21:46.02 | REdOG | anyone know what might cause a sip device to only work on for it's first call...additional calls only work after reboot/re-register |
| 21:46.22 | REdOG | it's like it looses it's audio but not its registration |
| 21:46.41 | WIMPy | NAT issues? |
| 21:46.46 | REdOG | local only |
| 21:46.48 | Kevin24 | SIP -> Opensips -> * 1.6 -> TE410P -> Telco -> PSTN |
| 21:47.52 | Kevin24 | it works when the call is coming from another asterisk caller, but from a Lucent Plexus-9000 we don't get it |
| 21:48.08 | Kevin24 | could it be the format? E164 on Lucent and on Asterisk another? |
| 21:48.41 | WIMPy | I guess you have to compare a successful and a failed one. |
| 21:49.30 | Kevin24 | Pamela Says: wow - all over - if we dont get ani - we may insert a NY or Miami number but we sell to everyone all over - i can try to get anis by customer - i would say US though woud be the most traffic |
| 21:49.32 | REdOG | yea, im mirroring up the port now |
| 21:51.05 | *** join/#asterisk The_REV (~The_REV@76-217-184-220.lightspeed.frsnca.sbcglobal.net) |
| 21:54.51 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 21:58.39 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 21:58.39 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 21:59.41 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 22:00.35 | *** part/#asterisk flyman_ (~flyman_@194.24.158.1) |
| 22:05.18 | jc319 | Did any of you guys P2V an asterisk box? & what software did you use? |
| 22:10.09 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 22:12.11 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 22:13.56 | jeffspeff | any suggestions on a cheap but good sip trunking services? |
| 22:15.54 | *** join/#asterisk JerJer (~jj@asterisk/original-h323-guy/JerJer) |
| 22:31.48 | voxio | jeffspeff: I've found voip.ms to be quite good (roughly 10 users on outbound calls at any one time) |
| 22:32.05 | voxio | they also do IAX failover on inbound calls |
| 22:34.56 | jeffspeff | voxio, thanks, i'll check them out |
| 22:35.57 | *** part/#asterisk Kevin24 (~kevin@190.99.204.24) |
| 22:39.46 | jc319 | yeah voip.ms is great |
| 22:45.19 | p3nguin | There is NO SUCH THING is SIP TRUNKING. Period. |
| 22:45.23 | *** join/#asterisk wonderworld (~ww@port-92-201-198-82.dynamic.qsc.de) |
| 22:45.26 | p3nguin | s/is/as/ |
| 22:45.30 | p3nguin | shit |
| 22:45.35 | WiretapWork | lol |
| 22:45.43 | p3nguin | It was better off before correction. |
| 22:46.13 | SunTsu | yelling is never good, no matter how much you correct it ;) |
| 22:46.33 | SunTsu | s/is never/never is/ |
| 22:48.11 | *** join/#asterisk gmaruzz (~gmaruzz@static-217-133-80-112.clienti.tiscali.it) |
| 22:48.21 | p3nguin | voxio: They also do SIP failover, or IAX2 primary. They don't specifically have "IAX2 failover." |
| 22:49.03 | p3nguin | You choose how to failover or what to do in the event that the first operation fails. |
| 22:56.10 | *** join/#asterisk celord (~celord@201.198.102.2) |
| 23:01.05 | voxio | ok well it fails over if the primary server is down |
| 23:01.09 | voxio | thats good enough for me |
| 23:19.09 | jkroon | what would it take to add arguments to the Goto() function, similar to Gosub()? |
| 23:19.39 | jkroon | situation: need to pass some additional info to a destination, not expecting the destination to ever return ... |
| 23:24.16 | carrar | variables? |
| 23:25.09 | WIMPy | That's why I'm still on macros. I dont like the idea of gosub without the intent to ever return. |
| 23:33.41 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 23:35.07 | jkroon | carrar, yea, but that incurs a few extra lines now of Set(), Set() ... |
| 23:41.12 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:42.21 | carrar | use a macro |
| 23:43.35 | *** join/#asterisk pdtpatrick (~pdtpatric@mainstwan.farheap.com) |
| 23:43.57 | jkroon | done. |
| 23:54.58 | *** join/#asterisk linuxplatform (~centoslin@88.87.48.55) |
