| 00:01.13 | *** join/#asterisk saftsack (n=saftsack@p57924321.dip.t-dialin.net) |
| 00:01.20 | saftsack | hi whats about t38 and *? |
| 00:02.12 | orpheee | well well...my Ivr do not accept playback and do not call julie you can see the error and extensions.conf there http://pastebin.com/d6460130e |
| 00:02.15 | orpheee | thanks |
| 00:03.08 | ruben23 | KyleK: iahve an asterisk server with public and local ip and also i wan to register SIP softphones from a remote location...how do i do the registration..? |
| 00:03.36 | ruben23 | i put the public ip of my asterisk on the softphone domain..its not working |
| 00:06.34 | jake[work] | <PROTECTED> |
| 00:07.35 | KyleK | odd |
| 00:07.50 | KyleK | ruben23: it works for me(tm) |
| 00:08.12 | ruben23 | KyleK:how did you do it..? |
| 00:08.19 | KyleK | are you sure the stuff in sip.conf is correct? |
| 00:08.43 | ruben23 | what are the config i have to set. |
| 00:08.59 | ruben23 | in particular |
| 00:10.48 | KyleK | http://pastebin.ca/1428020 |
| 00:10.51 | KyleK | nothing special |
| 00:16.12 | ruben23 | wow..same setting |
| 00:16.22 | ruben23 | but mine cannot connect |
| 00:16.45 | KyleK | look at sip traffic? |
| 00:17.03 | *** join/#asterisk nighty^ (n=nighty@210.188.173.245) |
| 00:18.58 | ruben23 | KyleK:i think my sip client is behind NAt |
| 00:20.36 | KyleK | oh, so add in a stun server on the client? |
| 00:21.58 | ruben23 | <PROTECTED> |
| 00:22.04 | KyleK | ~stun |
| 00:22.05 | infobot | somebody said stun was that feeling you get when you realise your SIP call actually got through!. Simple Traversal of UDP over NATs, or a client side method to cater to crappy sip servers, or a phaser setting |
| 00:23.13 | ruben23 | <PROTECTED> |
| 00:23.43 | KyleK | xlite has it |
| 00:23.54 | ruben23 | im using eyebeam.. |
| 00:24.01 | KyleK | ekiga does too, if others you'll have to look yourself |
| 00:30.26 | orpheee | <PROTECTED> |
| 00:40.27 | rhassing | orpheee, http://pastebin.com/d659e716e for one of my menus |
| 00:41.17 | orpheee | lol |
| 00:41.18 | *** join/#asterisk blkry (n=blkry@97.95.233.232) |
| 00:41.30 | rhassing | orpheee, ? |
| 00:42.05 | orpheee | don't understand your ivr .. too complex for me |
| 00:42.27 | rhassing | the first part is a loop, so it will only be played 3 times |
| 00:42.35 | rhassing | the second part is the menu |
| 00:47.28 | orpheee | but i don't why exten 1 is rejected |
| 00:51.18 | orpheee | yes but for that |
| 00:51.32 | orpheee | http://pastebin.com/d3e8dd234 |
| 00:55.23 | rhassing | maybe you should add an "exten => s,n,WaitExten(5)" just after s,1,Background(menu1) |
| 00:57.37 | orpheee | i try |
| 00:58.37 | orpheee | WOW man ! |
| 00:58.42 | orpheee | i try again lol |
| 01:05.01 | *** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
| 01:06.53 | telnettech | rhassing: trouble with ODBC was that on the remote server, the mysql service wasnt running correctly |
| 01:08.40 | telnettech | thought i would let you know with the help you gave me earlier |
| 01:08.51 | telnettech | im going home for the night!!!!!! |
| 01:13.15 | *** join/#asterisk ruben23 (n=AGENT@124.107.3.178) |
| 01:19.20 | orpheee | rhassing thanks, my ivr work ^^ |
| 01:19.47 | orpheee | but how can i do to choose directly my choice |
| 01:20.03 | orpheee | if i want directly choose "1" |
| 01:20.55 | *** join/#asterisk dwayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 01:23.18 | jplank | anyone in the UK that could show me their dial plan? |
| 01:23.33 | *** join/#asterisk emer08 (i=e@122.55.66.8) |
| 01:23.40 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 01:23.58 | emer08 | anyone who can help me in configuring unicall.conf? |
| 01:23.59 | emer08 | anyone who can help me in configuring unicall.conf? |
| 01:25.12 | *** join/#asterisk captiancrash (n=jonmoore@adsl-074-181-189-229.sip.owb.bellsouth.net) |
| 01:25.24 | emer08 | can anyone tell me how to configure the channels on unicall.conf to go both ways. incoming and outgoing |
| 01:25.33 | emer08 | bit 1 |
| 01:25.37 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:28.44 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-2ae817ed7dec4956) |
| 01:34.06 | KyleK | can I force an order on matching extension.conf rules? |
| 01:35.26 | [TK]D-Fender | KyleK: Yes, by grouping the extens in different "include"-ed contexts |
| 01:35.45 | [TK]D-Fender | KyleK: Otherwise extens within a flat context follow a specific sort order |
| 01:36.20 | [TK]D-Fender | KyleK: Though generally more specific extens match before patterns (just due to raw alphanumberic ASCII sorting |
| 01:36.21 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.6) |
| 01:37.06 | KyleK | yea its the raw alphanumeric sorting that was trouble for me |
| 01:39.06 | *** join/#asterisk JT_ (n=j@unaffiliated/jt) |
| 01:39.20 | [TK]D-Fender | KyleK: there is a WIKI page which describes the ordering between sorting within a context and how bas extens in a contex are processed relative to includes |
| 01:39.44 | KyleK | the voip-info wiki? |
| 01:40.19 | [TK]D-Fender | KyleK: Yes |
| 01:40.42 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 02:04.27 | *** join/#asterisk tobias (n=tobias@user-0ce2hp1.cable.mindspring.com) |
| 02:11.58 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 02:14.09 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:18.24 | emer08 | can anyone tell me how to configure the channels on unicall.conf to go both ways. incoming and outgoing |
| 02:25.38 | *** join/#asterisk chendy (n=chatzill@58.251.102.216) |
| 02:29.29 | *** join/#asterisk trentcreek (n=kvirc@200.94.227.117) |
| 02:37.29 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 02:38.22 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 02:42.11 | *** join/#asterisk voxter (n=voxter@76.77.91.251) |
| 02:48.37 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 03:01.15 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-618264f9e1e1e1dd) |
| 03:02.29 | *** join/#asterisk DarkLogik (n=darklogi@76.73.51.195) |
| 03:11.09 | KyleK | is there a NULL channel i can dial? |
| 03:11.23 | ruben23 | hi |
| 03:11.51 | ruben23 | can i use my TDM400P as my timing device for mya asterisk server. |
| 03:12.27 | ruben23 | how do i installed it...but i already install my zaptel driver |
| 03:13.15 | KyleK | sounds like you can use it, maybe it defaults to it? |
| 03:14.48 | jaytee | if you have zaptel hardware and a zaptel driver loaded for your TDM400P then Asterisk will take timing for apps like MeetMe from it. |
| 03:15.22 | jaytee | if you installed Zaptel first before installing Asterisk |
| 03:15.29 | ruben23 | yes |
| 03:15.41 | ruben23 | actually i ahve a workign asterisk server |
| 03:15.51 | ruben23 | but i want to add my TDM400P |
| 03:15.57 | ruben23 | as my timing device |
| 03:17.08 | ruben23 | so i just add it up |
| 03:17.10 | jaytee | then install the card, reinstall and reconfigure zaptel for the card and then recompile asterisk with zaptel enabled |
| 03:17.12 | ruben23 | then let it run |
| 03:22.18 | ruben23 | jaytee:my TDM400P card have no modules |
| 03:22.25 | ruben23 | is it ok..? |
| 03:23.25 | KyleK | probably |
| 03:24.15 | jaytee | hmmmm? no modules? what kind of bargain basement operation are you running there? |
| 03:25.15 | ruben23 | jaytee: what i have here is the plane TDM400P wildcard only no module,,,can i still used this as timing device..? |
| 03:25.48 | KyleK | give it a shot since you have the card |
| 03:25.57 | jaytee | ruben23, I'd imagine so but not sure. |
| 03:26.09 | ruben23 | ok ill try it... |
| 03:26.35 | jaytee | if the driver will load for it but you have no actual channels it should still work. |
| 03:27.00 | ruben23 | how do i chweck the driver for this..? |
| 03:27.07 | ruben23 | if it loaded |
| 03:27.27 | jaytee | ps aux | grep zaptel |
| 03:29.05 | ruben23 | this is a production server im quite nervous installing this.... |
| 03:30.49 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 03:32.52 | jaytee | you're kidding, right? you're learning asterisk and you're trying to install on a production system? are you nuts? |
| 03:33.36 | KyleK | back the config up? |
| 03:34.08 | rob0 | Can't find reverse gear, how can I back it up?!?! |
| 03:40.34 | [TK]D-Fender | ruben23: And I've answered this question a half-doen times. |
| 03:42.37 | [TK]D-Fender | rob0: Don't forget your solder and sound cards! You'll miss the beeping sounds as you back it up ;) |
| 03:43.20 | ruben23 | :-( |
| 03:44.43 | *** join/#asterisk mortsmel (n=nnscript@corp-nat.kaslnet.net) |
| 03:45.52 | mortsmel | trying to determin why my trixbox completely ignores advanced portion of voicemail and immediately goes to the requesting for previous message (by pressing star) |
| 03:46.30 | jaytee | "roger that, Houston. ISS crew report that his ship's starboard thruster assembly has been sheared off in a collision with one of the solar panel assemblies." |
| 03:46.59 | mortsmel | wha |
| 03:47.09 | mortsmel | on hubble? |
| 03:47.57 | jaytee | nope, talking about the Good Ship Ruben23 which is in a slightly deranged orbit at the moment. |
| 03:49.22 | *** join/#asterisk Reality-X (n=rx2@ip68-97-143-81.ok.ok.cox.net) |
| 03:49.28 | Reality-X | sup folks |
| 03:49.42 | jaytee | sup? |
| 03:50.08 | [TK]D-Fender | jaytee: Unidentified FAIL Object :) |
| 03:50.21 | jaytee | hehehe |
| 03:50.23 | mortsmel | ah |
| 03:50.29 | Reality-X | just checking, is there a newer version of the php agi libs than version 2.14, or what libs would you all recommend for php agi development |
| 03:50.49 | mortsmel | thinks a quick fix is just reinstalling trixbox |
| 03:50.49 | [TK]D-Fender | Reality-X: What do you see on the dev's site? |
| 03:50.58 | mortsmel | sucks ... got 30 customers on it though :( |
| 03:51.24 | drmessano | or getting rid of trixbox |
| 03:51.34 | mortsmel | and replacing it with |
| 03:51.35 | jaytee | I was gonna say... |
| 03:51.43 | Reality-X | 2.14 on sf, but im just checking to see if that is the "one" that most folks use |
| 03:52.03 | drmessano | One of the 10 other PBX distros based on a Asterisk and FreePBX |
| 03:52.17 | drmessano | Do some research |
| 03:52.31 | mortsmel | aye aye |
| 03:53.14 | drmessano | Thats what I always say.. an aye for an aye |
| 03:53.42 | jaytee | "second star to the left and clear on till morning" |
| 03:54.59 | rob0 | ewww, paying customers ... on trixbox?!? |
| 03:57.31 | mortsmel | just looking for a freepbx / asterisk combo only now ... w/o all the extra bloat :) |
| 03:57.36 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 03:58.16 | drmessano | AsteriskNOW or PBX In a Flash are my recommendations |
| 03:58.17 | Qwell | mortsmel: AsteriskNOW |
| 03:58.28 | Qwell | Whoever wrote that should get tons of money and free beers. |
| 03:58.40 | mortsmel | and does it interface well w/ metaswitches? |
| 03:58.57 | mortsmel | via sip trunking |
| 03:58.59 | mortsmel | bleh |
| 03:59.03 | drmessano | ..... |
| 03:59.06 | mortsmel | i'm asking stupid questions of course it does |
| 04:00.48 | mortsmel | interfacing w/ a friends meta right now ... 1 primary sip trunk and my asterisk box is basically terminating my customers ... |
| 04:02.38 | drmessano | Youa lready mentioned you had Trixbox.. the responses given were based on your current setup, which is Asterisk + GUI (which isnt technically FreePBX anymore in trixbox, its a fork) |
| 04:02.49 | drmessano | So yea, stupid question |
| 04:11.49 | Juggie | is there a way to print all the variables on a channel |
| 04:15.48 | [TK]D-Fender | Juggie: "core show channel [channel] |
| 04:17.37 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 04:18.51 | Juggie | any dialplan function that does it? |
| 04:18.59 | Juggie | so i could noop it out |
| 04:19.22 | *** part/#asterisk M1s3ry (n=jbigbee@boromir.api-digital.com) |
| 04:19.25 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 04:23.26 | *** join/#asterisk trentcreek (n=kvirc@200.94.227.117) |
| 04:24.39 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 04:32.30 | *** join/#asterisk nephfl (n=none@wsip-98-175-64-147.ga.at.cox.net) |
| 04:43.51 | nephfl | hello |
| 04:44.57 | nephfl | anyone up? |
| 04:45.11 | jake[work] | yes |
| 04:46.25 | nephfl | i have a problem with my asterisk install..it doesnt seem to be loading all the modules, when i restart it says it cant load all the features, could that mean too little memory? |
| 04:46.50 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 04:46.59 | jake[work] | can you pastebin the output? |
| 04:47.23 | jake[work] | should be error messages. start asterisk -vvvvgcd |
| 04:47.26 | nephfl | output of what? problem is i cant get sip to connect and dont see any sip module options in cli |
| 04:47.54 | *** join/#asterisk MrNaz (n=mrnaz@ppp121-44-203-184.lns10.mel4.internode.on.net) |
| 04:48.07 | jake[work] | output of: asterisk -vvvvgcd |
| 04:48.10 | [TK]D-Fender | nephfl: You are clearly getting SOME kind of message so you'd better show us |
| 04:50.30 | trentcreek | Maybe the same problem I had of incorrect directory |
| 04:50.41 | jake[work] | maybe :) |
| 04:51.55 | nephfl | that was alot of crap |
| 04:52.02 | jake[work] | but let's not jump to conclusions before we see the output |
| 04:52.38 | nephfl | but it loaded properly that time |
| 04:52.39 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.17) |
| 04:53.19 | jake[work] | how was that different than what you did before? |
| 04:54.36 | nephfl | dunno |
| 04:54.52 | jake[work] | great! |
| 04:54.54 | nephfl | this system is running in an OpenVZ container |
| 04:55.05 | jake[work] | how did you start asterisk last time? |
| 04:55.30 | nephfl | just started it..no arguments |
| 04:55.35 | jake[work] | ok |
| 04:57.24 | trentcreek | What in asterisk is producing these headers? http://pastebin.ca/1428153 |
| 04:57.34 | jake[work] | i don't think the options i gave you should've been any different. maybe somebody else can comment |
| 04:59.35 | trentcreek | the "Unknown" I need to change because I am getting denied authentication |
| 05:09.48 | nephfl | about how much memory is require for *1.6? |
| 05:11.36 | trentcreek | maybe 50MB |
| 05:12.45 | trentcreek | but as you have it do more things, it goes up quickly |
| 05:14.34 | trentcreek | just for making calls, you would probably only need 128MB..which includes the OS |
| 05:28.51 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 05:29.16 | *** join/#asterisk mbuf (n=user@61.16.248.242) |
| 05:33.04 | *** join/#asterisk frantic667 (n=toffifee@dsbg-4db5cd4f.pool.einsundeins.de) |
| 05:33.10 | frantic667 | hello there |
| 05:33.30 | mbuf | if i don't want asterisk to use postgresql, where do I disable it? |
| 05:33.56 | mbuf | ERROR[3591]: res_config_pgsql.c:961 pgsql_reconnect: PostgreSQL RealTime: Failed to connect database asterisk on 127.0.0.1: |
| 05:33.56 | mbuf | <PROTECTED> |
| 05:34.49 | rob0 | Hmmm, how/why did you enable it in the first place? |
| 05:35.22 | mbuf | rob0: i have no idea! newbie here; just using the defaults from Fedora 10 asterisk-1.6 installation |
| 05:35.36 | mbuf | rob0: let me paste the logs to pastebin.ca |
| 05:35.41 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 05:35.50 | rob0 | ~book |
| 05:35.51 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 05:36.13 | frantic667 | caller does not hear a dialtone when calling my asterisk, is there any setting to enable? everything works, asterisk even answers the call... but until the answering ther is silence on the line |
| 05:37.23 | mbuf | rob0: http://pastebin.ca/1428174 ; i read through the book; i was looking for a simple start-to-finish HOWTO to test SIP |
| 05:40.57 | trentcreek | mbuf: www.voip-info.org has them |
| 05:43.30 | mbuf | trentcreek: thanks, i am looking into it; |
| 05:44.20 | frantic667 | okay, my problem was solved by itself... now i have a dialtone.... strange... |
| 05:48.03 | *** join/#asterisk CrazyTux (n=brandon@ip68-4-117-195.oc.oc.cox.net) |
| 05:49.07 | *** join/#asterisk stijnbe (n=stijnbe@78-22-110-114.access.telenet.be) |
| 05:49.15 | *** part/#asterisk CrazyTux (n=brandon@ip68-4-117-195.oc.oc.cox.net) |
| 05:51.42 | *** join/#asterisk CrazyTux (n=brandon@ip68-4-117-195.oc.oc.cox.net) |
| 05:53.35 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-bb7939080b3a1042) |
| 06:04.30 | mbuf | ok, i renamed res_ldap.conf, res_pgsql.conf, oss.conf to their respective .sample files, and now I get asterisk CLI |
| 06:08.02 | trentcreek | groovy |
| 06:12.43 | *** join/#asterisk arthax0r (i=arthax0r@gateway/shell/blinkenshell.org/x-88d979721e6df54d) |
| 06:14.00 | *** join/#asterisk oej_ (n=olle@ns.webway.se) |
| 06:15.14 | *** join/#asterisk denysonique (n=dennis@unaffiliated/dennisonicc) |
| 06:15.16 | denysonique | Hi |
| 06:15.55 | denysonique | My asterisk is unable to send anything to the SIP client which is on my pc behind nat |
| 06:16.08 | denysonique | I have nat=yes and qualify=yes set |
| 06:17.12 | trentcreek | denysonique: make sure the client registers |
| 06:17.33 | trentcreek | else it will not receive a call under NAT |
| 06:17.43 | denysonique | trentcreek: it tries to but it fails with a 408 |
| 06:17.50 | denysonique | because it gets on response |
| 06:18.10 | denysonique | s/on/no |
| 06:18.20 | trentcreek | which is behind NAT? |
| 06:18.25 | denysonique | the client |
| 06:18.30 | denysonique | yes |
| 06:18.44 | trentcreek | the client fails to register? |
| 06:18.52 | denysonique | yes |
| 06:19.14 | trentcreek | do you see it trying to register on the server? |
| 06:21.13 | denysonique | trentcreek: http://rafb.net/p/q5Aqbe80.html |
| 06:21.17 | denysonique | yes |
| 06:22.25 | *** join/#asterisk s0lid (n=s0lid@210.213.254.49) |
| 06:22.46 | trentcreek | denysonique: better check your settings on both sides. Either bad password, or extension, or both |
| 06:22.51 | trentcreek | I think |
| 06:23.01 | denysonique | hmm |
| 06:23.16 | denysonique | what is the contact section? |
| 06:23.20 | denysonique | in that log? |
| 06:23.39 | denysonique | because it refers to the other nic on the server |
| 06:23.50 | denysonique | that has no connection with the client |
| 06:24.48 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 06:25.02 | denysonique | trentcreek: I haven't edited extensions.conf yet |
| 06:25.32 | trentcreek | yeah..so you better do that...it is saying the client is not authorized to log in |
| 06:29.15 | denysonique | trentcreek: I have edited the extensions.conf but it still doesn't work |
| 06:29.28 | trentcreek | did you restart? |
| 06:29.29 | denysonique | the client doesn't even receive 401 from the server |
| 06:29.34 | trentcreek | i mean reload |
| 06:29.37 | denysonique | trentcreek: I have reloaded |
| 06:29.54 | denysonique | using the asterisks cli |
| 06:30.44 | trentcreek | tell you what...go on youtube...and look up "Episode 5¨" it is a video on how to setup...dont waste time with that one..go to the one on the link....they also have the example code they used in the video |
| 06:36.54 | *** join/#asterisk xpot (n=james@70.91.210.233) |
| 06:36.56 | *** join/#asterisk xrmx__ (n=rm@host119-200-dynamic.180-80-r.retail.telecomitalia.it) |
| 06:43.02 | *** join/#asterisk mikkel (n=mikkel@130.226.37.126) |
| 06:44.13 | denysonique | failed for my ip peer is not supposed to register |
| 06:46.19 | denysonique | [May 20 08:45:54] NOTICE[3141]: chan_sip.c:15236 handle_request_register: Registration from '<sip:dennis@tatuacy.com>' failed for '86.138.27.142' - Peer is not supposed to register |
| 06:47.23 | frantic667 | did you set host=dynamic? |
| 06:48.15 | denysonique | frantic667: I set host to the IP address of the server |
| 06:49.07 | trentcreek | did yo follow the video? |
| 06:49.23 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 06:49.26 | frantic667 | denysonique: the server ip is static? no hostnames set wich could be resolved to another ip? |
| 06:49.27 | denysonique | trentcreek: the video is unclear |
| 06:49.42 | trentcreek | just look at the samples to download |
| 06:50.14 | denysonique | frantic667: it is static |
| 06:50.20 | denysonique | the IP of asterisk |
| 06:50.52 | trentcreek | those samples are the most easiest to understand how to set it up |
| 06:51.49 | denysonique | trentcreek: my sip.conf is very simple and should work |
| 06:52.09 | trentcreek | just verify with the samples |
| 06:52.36 | trentcreek | the first time I got asterisk..i used those samples ans was up and running in minutes |
| 06:52.41 | *** join/#asterisk mikkel (n=mikkel@130.226.37.126) |
| 06:55.12 | denysonique | trentcreek: they are inappropriate for my configuration |
| 06:55.19 | denysonique | there is no nat |
| 06:55.53 | trentcreek | you sais the server was not |
| 06:55.59 | trentcreek | but the clients are? |
| 06:56.10 | denysonique | yes |
| 06:56.26 | trentcreek | so the samples are fine. |
| 06:56.32 | trentcreek | your client will be DYNAMIC |
| 06:58.27 | denysonique | if the client is behind nat I need to specify nat=yes then |
| 06:59.59 | trentcreek | yes |
| 07:00.14 | denysonique | but why it still doesn't work? |
| 07:01.01 | *** join/#asterisk unasi7 (n=unasi7@84-75-23-151.dclient.hispeed.ch) |
| 07:02.03 | unasi7 | question: when i place a register in sip.conf, which context it will serach after the extension in extensions.conf (i always get : extension not found)? |
| 07:03.36 | trentcreek | i dont know |
| 07:06.53 | frantic667 | unasi7: when I understand right: did you define a name for the register with a "/" behind the registrar? |
| 07:07.24 | frantic667 | eg: login:secret@sip.provider.com/login |
| 07:07.35 | frantic667 | then the extension "login" will be used |
| 07:09.48 | *** join/#asterisk Eberx (n=Eberx@203.201.181.17) |
| 07:09.53 | Eberx | Hi All |
| 07:09.57 | unasi7 | frantic667, yes. i do. i have a extension 1670 in sip.conf register |
| 07:09.57 | denysonique | hurray it works! |
| 07:10.12 | unasi7 | now it works.... it will search in [local] for it |
| 07:10.20 | Eberx | Does asterisk can work like SIP Proxy ? |
| 07:10.27 | denysonique | Eberx: yes |
| 07:10.42 | denysonique | I just had to specify bindaddr |
| 07:10.50 | denysonique | because the server has 2 nics |
| 07:11.21 | Eberx | means It can register customer to directly itself |
| 07:11.44 | denysonique | Eberx: ? |
| 07:12.11 | Eberx | asterisk can register customer itself |
| 07:12.17 | Eberx | ? |
| 07:15.58 | *** join/#asterisk rhassing_work (n=rob_work@ti152.telin.nl) |
| 07:17.23 | denysonique | Eberx: I don't undertand your question |
| 07:17.34 | *** join/#asterisk stijnbe (n=stijnbe@router.begen1.office.netnoc.eu) |
| 07:26.33 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 07:29.27 | *** join/#asterisk krdian (i=krdian@killer.radom.net) |
| 07:29.32 | krdian | hi |
| 07:29.48 | KyleK | hi |
| 07:30.02 | rhassing_work | hi |
| 07:31.17 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 07:34.25 | *** join/#asterisk frantic667 (n=toffifee@dsbg-4db5f8fd.pool.einsundeins.de) |
| 07:37.25 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 07:42.24 | *** join/#asterisk ck_28 (n=CK@212.98.141.199) |
| 07:42.31 | ck_28 | hi ppl |
| 07:42.43 | ck_28 | on asterik cli |
| 07:42.47 | ck_28 | fax show stats |
| 07:42.50 | ck_28 | Digium T.38 |
| 07:42.51 | ck_28 | Licensed Channels : 1 |
| 07:42.51 | ck_28 | Max Concurrent : 1 |
| 07:42.51 | ck_28 | Success : 4 |
| 07:42.51 | ck_28 | Canceled : 0 |
| 07:42.51 | ck_28 | No Fax : 0 |
| 07:42.53 | ck_28 | Partial : 1 |
| 07:42.55 | ck_28 | Negotiation Failed : 0 |
| 07:42.57 | ck_28 | Train Failure : 2 |
| 07:42.59 | ck_28 | Protocol Error : 0 |
| 07:43.01 | ck_28 | IO Partial : 0 |
| 07:43.03 | ck_28 | IO Fail : 0 |
| 07:43.07 | ck_28 | what do Train Failure means |
| 07:43.26 | ck_28 | from where i can get a detailed information for this stats |
| 07:49.30 | tzafrir_laptop | ~pb |
| 07:49.31 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 07:49.41 | tzafrir_laptop | (for next time) |
| 07:54.03 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 07:55.36 | *** join/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 07:56.44 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 07:56.52 | ck_28 | ok any idea ? |
| 07:57.12 | ck_28 | from where i can get the document which explain these parameters |
| 08:02.51 | *** join/#asterisk TEXYNZ (n=TEX@124-197-26-124.callplus.net.nz) |
| 08:03.13 | TEXYNZ | I having a few issues with external SIP extensions |
| 08:03.26 | TEXYNZ | can someone shed some light on my problem |
| 08:03.39 | *** join/#asterisk ctp (n=ctp@brsg-d9befcb8.pool.mediaWays.net) |
| 08:03.45 | TEXYNZ | internal IP of asterisk box in sip client works |
| 08:03.54 | TEXYNZ | but externals ip of asterisk box doesn't |
| 08:04.02 | TEXYNZ | box in DMZ |
| 08:07.26 | *** join/#asterisk ctp (n=ctp@brsg-d9befcb8.pool.mediaWays.net) |
| 08:11.03 | *** part/#asterisk Flyser (n=Flyser@unaffiliated/flyser) |
| 08:12.23 | krdian | ck_28: connectivity or connection problems i think, probably too poor quality connection |
| 08:13.18 | krdian | is back. and if you laugh, i'll kill you! |
| 08:23.36 | *** join/#asterisk oej (n=olle@fw01d.snowmen.se) |
| 08:24.17 | ck_28 | krdian thanks,kindly from where you get this result |
| 08:25.35 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 08:26.37 | Eberx | Does asterisk work with MySQL backend ? |
| 08:28.40 | krdian | ck_28: from my own experience :) did u try to find answer to your problem on http://www.digium.com/en/supportcenter/ ? |
| 08:36.15 | rhassing_work | Eberx, yes it does |
| 08:37.05 | ck_28 | krdian thanks |
| 08:37.09 | Eberx | rhassing_work, that means sip users information located in mysql |
| 08:37.30 | ck_28 | krdian i have no support account on support center |
| 08:38.00 | rhassing_work | Eberx, It is possible, but is pretty complicated |
| 08:38.09 | *** join/#asterisk fors1 (n=forsen@pat-tdc.opera.com) |
| 08:38.19 | Eberx | rhassing_work, I see |
| 08:39.08 | krdian | ck_28: just register new account then you can get access to docs |
| 08:39.28 | krdian | ck_28: its free |
| 08:39.34 | Eberx | rhassing_work, If asterisk have 1000 sip customers then how to manage that customers. put the all information on sip.conf |
| 08:41.03 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:42.31 | denysonique | can someone try to register to my server and dial my extension? |
| 08:42.35 | denysonique | Please |
| 08:43.27 | frantic667 | i have a damn slow connection here, but i can try |
| 08:45.23 | rhassing_work | Eberx, http://www.voip-info.org/wiki/view/Asterisk+at+large Maybe you should consider using OpenSer as well |
| 08:45.38 | Eberx | I see |
| 08:45.51 | Eberx | rhassing_work, thank you |
| 08:46.03 | rhassing_work | Eberx, NP, you're welcome |
| 08:49.46 | fors1 | aoeu/win 36 |
| 08:50.02 | denysonique | when I try to dial an extension 1234 that should connect me with another user I get a 407 error on the dialing client |
| 08:50.32 | denysonique | when the two users user a and b are registred on the same softphone the extension works |
| 08:50.44 | denysonique | a dials 1234 and it connects to b |
| 08:51.26 | denysonique | but when the user b is registered somewhere else I am unable to dial it |
| 08:56.02 | *** join/#asterisk nicola_pav (n=chatzill@83.244.78.241) |
| 08:56.25 | nicola_pav | hello |
| 08:56.35 | nicola_pav | have spa3102 and asterisk |
| 08:56.47 | nicola_pav | i managed to register both ports, fxo and fxs |
| 08:56.47 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 08:56.55 | *** join/#asterisk voxter (n=voxter@76.77.91.251) |
| 08:56.56 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 08:56.59 | nicola_pav | i have also xlite |
| 08:57.11 | nicola_pav | when i call from xlite to the fxs |
| 08:57.21 | nicola_pav | i want the call to be forwarded to the fxo |
| 08:57.27 | nicola_pav | how can i do that? |
| 08:58.06 | nicola_pav | i can call the fxs from xlite successfully |
| 09:00.02 | *** join/#asterisk gregd (n=gregd@80-41-192-81.dynamic.dsl.as9105.com) |
| 09:02.38 | *** join/#asterisk _pepo_ (n=pepo@200.55.224.2) |
| 09:02.43 | _pepo_ | hi friends |
| 09:08.16 | _pepo_ | I am using asterisk in my xserver with an Intel Xeon quad core, but I can only handle 150 concurrent SIP calls ... Is there some way to improve and handle more SIP calls? |
| 09:08.35 | *** join/#asterisk fors1 (n=forsen@pat-tdc.opera.com) |
| 09:17.25 | rhassing_work | I have some strange problems with my voipbuster account and dnsmgr... http://pastebin.com/d52da1e40 |
| 09:18.24 | mvanbaak | _pepo_: couple of things to check: bandwidth, transcoding, load of the box |
| 09:18.34 | denysonique | _pepo_: kill x :) |
| 09:20.41 | _pepo_ | The bandwidth is Fastethernet/Gigaethernet, I am using gsm/ulaw and there is nothing more than asterisk... what do you think? |
| 09:21.09 | *** join/#asterisk SebastianS (n=schu@dsl-static-111.212-5-200.telecom.sk) |
| 09:21.26 | tzafrir_laptop | nicola_pav, it seems you basically need to translate that to asterisk-speak . Dial(SIP/peer-of-spa3102-fxo) |
| 09:21.29 | tzafrir_laptop | or: |
| 09:21.41 | tzafrir_laptop | Dial(SIP/peer-of-spa3102-fxo/number) |
| 09:22.04 | nicola_pav | tzafrir_laptop: we talekd yesterday :) |
| 09:24.13 | tzafrir_laptop | nicola_pav, please show relevant trace from the asterisk CLI |
| 09:24.15 | tzafrir_laptop | ~pb |
| 09:24.16 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 09:25.22 | nicola_pav | i can call successfully the fxs on the spa from an xlite |
| 09:25.33 | nicola_pav | under the tab "User 1" |
| 09:25.50 | nicola_pav | when no answer, i am trying to forward the call to the fxo |
| 09:26.32 | nicola_pav | i put as dial plan: (0<:@gw0>) since i am behind a PBX |
| 09:27.11 | nicola_pav | NOTICE[30839]: chan_local.c:526 local_alloc: No such extension/context (105s0<:@default creating local channel |
| 09:27.12 | nicola_pav | May 20 12:18:17 NOTICE[30839]: app_dial.c:481 wait_for_answer: Unable to create local channel for call forward to 'Local/(105s0<:@default' (cause = 0) |
| 09:27.14 | nicola_pav | <PROTECTED> |
| 09:27.16 | nicola_pav | <PROTECTED> |
| 09:27.26 | nicola_pav | above is what i get from asterisk cli |
| 09:28.02 | tzafrir_laptop | infobot, tell nicola_pav about pb |
| 09:29.14 | tzafrir_laptop | that's the place to paste things of more than three lines, to avoid cloberring the channel ... |
| 09:30.41 | nicola_pav | i paste what i want in one of the links u gave me and what after? |
| 09:32.23 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
| 09:34.06 | frantic667 | why is my souds language suddenly english? it was german until now and i did not change anything affecting the language settings (imho). is this a bug? |
| 09:36.42 | jchillerup | frantic667: have you dist-upgraded or done something that could change your locale+ |
| 09:38.38 | nicola_pav | # May 20 12:33:41 NOTICE[30884]: chan_local.c:526 local_alloc: No such extension/context (105s0<:@default creating local channel # May 20 12:33:41 NOTICE[30884]: app_dial.c:481 wait_for_answer: Unable to create local channel for call forward to 'Local/(105s0<:@default' (cause = 0) # == Everyone is busy/congested at this time (1:0/0/1) # == Auto fallthrough, channel 'SIP/xlite2-09b4c8e8'... |
| 09:38.40 | nicola_pav | ...status is 'CHANUNAVAIL' |
| 09:39.27 | frantic667 | jchillerup: no, just commented-out an extension (an unused one) and reloaded sip. if i call echotest internal it is still german, but when i call from outside the voicemailbox speaks enlish... |
| 09:39.30 | nicola_pav | tzafrir_laptop: please c the output above |
| 09:39.48 | jchillerup | frantic667: I'm sorry then, I can't help you |
| 09:40.09 | frantic667 | jchillerup: hmmm, thanky, anyway :-) |
| 09:40.14 | frantic667 | *thanks |
| 09:40.41 | tzafrir_laptop | nicola_pav, you paste in such a pastebin, and it creates a page. post here a link to that page |
| 09:42.07 | nicola_pav | http://pastebin.ca/1428430 |
| 09:42.21 | *** part/#asterisk jchillerup (n=aaa@hald.gbar.dtu.dk) |
| 09:43.56 | nicola_pav | tzafrir_laptop: http://pastebin.ca/1428430 |
| 09:44.58 | tzafrir_laptop | (105s0<: - straneg name for an extension |
| 09:45.38 | nicola_pav | my pstn line is behind a PBS |
| 09:45.41 | nicola_pav | PBX* |
| 09:45.51 | tzafrir_laptop | btw: using "default" as the context is something you should generally avoid. Certainly if you'll have your system open to external voip providers |
| 09:45.53 | nicola_pav | there is phone here that has the extension 105 |
| 09:45.58 | nicola_pav | i want to call it |
| 09:46.20 | tzafrir_laptop | can you pastebin the relevant parts of your dialpla (extensions.conf) ? |
| 09:47.53 | nicola_pav | http://pastebin.ca/1428440 |
| 09:54.31 | frantic667 | only way to solve my problem seemed to be rewriting the enlish sound files with the german ones... not very clean, but works in my case |
| 09:56.47 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
| 10:02.25 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 10:02.49 | nicola_pav | tzafrir_laptop: extensions.conf http://pastebin.ca/1428440 |
| 10:04.55 | *** join/#asterisk ceegee (n=christia@mail.cg-networks.de) |
| 10:04.58 | ceegee | hello |
| 10:06.00 | *** join/#asterisk ck_28 (n=CK@212.98.141.199) |
| 10:06.09 | ceegee | this is a little bit offtopic, but I am trying out a freepbx function named callerid lookup source with an http source, I configured it but nothing happens, even in httpd access.log nothing happens |
| 10:07.08 | ceegee | I dont know where to look at for debug information |
| 10:07.08 | ck_28 | *****i am using asterisk free fax with capabilities T38 and G711 how to force using T38 all the way ?******** |
| 10:17.27 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 10:18.12 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
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| 11:12.40 | *** join/#asterisk ceegee (n=christia@mail.cg-networks.de) [NETSPLIT VICTIM] |
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| 11:55.04 | frantic667 | someone an idea why my asterisk does not react to DTMF tones while Backgound(vm-intro)? When pressing "1" it simply goes on without executing the extension "1", nothing is showed in the CLI |
| 11:55.08 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 11:59.51 | wdoekes | wrong dtmfmode? |
| 12:00.43 | frantic667 | I set in the indications.conf country=de, do I need more settings? |
| 12:04.27 | frantic667 | okay, adding dtmfmode=rfc2833 to sip.conf was the solution, thanks for the hint! |
| 12:06.20 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
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| 12:12.38 | *** join/#asterisk SuD (n=Ask@89.140.32.2.static.user.ono.com) |
| 12:13.13 | SuD | hi, i need to set up a B410P ISDN card with a recent kernel |
| 12:13.40 | SuD | misdn 1.x not supported, misdn 2.x errors and/or crashes, dahdi doesn't support TE+PTMP, ... any advice? |
| 12:15.15 | SuD | http://rafb.net/p/3Hl1a750.html <-- misdn v2 error |
| 12:16.45 | tzafrir_laptop | SuD, what kernel do you have? |
| 12:17.01 | *** join/#asterisk alinuxd555555556 (n=alinux@193.227.191.90) |
| 12:17.08 | tzafrir_laptop | SuD, dahdi does support ptmp te |
| 12:17.23 | tzafrir_laptop | it does not support ptmp nt, however |
| 12:17.33 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 12:18.43 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:18.54 | alinuxd555555556 | hi all..this is slightly offtopic..I have 6 linksys adapters..and I need a provider to allow me set them all up on the same account..and for an incoming number that sends incomding calls to any of thse VOIP adapters..I know it can be done on asterisk...so I am looking for a provider who can provide this. thanks |
| 12:19.25 | SuD | sorry about the mistake, 2.6.28-11-server (ubuntu server 9.04) |
| 12:19.38 | [TK]D-Fender | alinuxd555555556: Any provider that runs a hosted PBX solution might offer this. Go shop around |
| 12:23.21 | *** join/#asterisk Dovid (n=annon@ool-4355e297.dyn.optonline.net) |
| 12:23.29 | Dovid | hi. is the TE110P still in production ? |
| 12:25.27 | [TK]D-Fender | Dovid: http://www.digium.com/en/products/digital/ |
| 12:25.41 | [TK]D-Fender | Dovid: Looks liek otherwise they'd ahve pulled it from their list |
| 12:27.19 | Dovid | thanks TK. I don't see it there so I assume its out of production. if it was out i was trying to see when it went out |
| 12:27.23 | Dovid | thanks for the URL |
| 12:29.21 | [TK]D-Fender | Dovid: the 120 added VoiceBus |
| 12:29.38 | [TK]D-Fender | (or somthing ont he PCI side...) |
| 12:29.51 | [TK]D-Fender | Porbably mixing the tech part of that up... |
| 12:29.53 | [TK]D-Fender | asjdhasjdf |
| 12:29.54 | [TK]D-Fender | gah. |
| 12:30.40 | Dovid | lol |
| 12:37.04 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:39.11 | *** join/#asterisk jmacz (n=jmacz@200.75.94.174) |
| 12:43.41 | *** join/#asterisk Orbixx (n=Orbixx@office.exoware.net) |
| 12:44.22 | *** join/#asterisk HenrikBe (n=zapphir@h204n4fls32o954.telia.com) |
| 12:44.30 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:44.36 | HenrikBe | is there anyone who has experience with ajam? |
| 12:45.24 | HenrikBe | I need to see the status (logged in or not) of a specific agent through a http request |
| 12:47.31 | *** join/#asterisk qdk (n=qdk@81.7.168.130) |
| 12:48.41 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 12:59.39 | *** join/#asterisk Nilzao (n=nils@200-168-146-103.dsl.telesp.net.br) |
| 12:59.49 | Nilzao | sup hackers |
| 13:00.12 | Nilzao | i will jornal my tests here |
| 13:00.24 | eppigy | say what |
| 13:00.41 | Nilzao | if someone know the path, help will be apreciated |
| 13:00.58 | Nilzao | installed ubuntu server 9.4 + asterisk 1.6.1 |
| 13:01.01 | Nilzao | without gui |
| 13:01.11 | Nilzao | have 2 fxo wildcard clones installed |
| 13:01.14 | Nilzao | started now |
| 13:02.04 | *** join/#asterisk esaym (n=user@cpe-24-174-186-34.satx.res.rr.com) |
| 13:02.53 | SuD | misdn git head version seems to work with kernel 2.6.28, nice... |
| 13:03.47 | jaytee | TRABAJO |
| 13:04.33 | eppigy | DONDE ESTA |
| 13:04.56 | beek | morning jaytee |
| 13:05.02 | SuD | que dicen? |
| 13:05.08 | jaytee | morning beek |
| 13:05.50 | Nilzao | edited the /etc/dahdi/modules and comented all the modules, leaving the wcfxo |
| 13:11.16 | *** join/#asterisk captiancrash (n=captianc@70.159.118.70) |
| 13:14.29 | *** join/#asterisk ghenry (n=ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 13:14.35 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
| 13:15.03 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 13:18.41 | *** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com) |
| 13:19.20 | Nilzao | edited /etc/asterisk/dahdi_cfg |
| 13:20.07 | Nilzao | dahdi_cfg -vvvvv worked and list the 2 fxo cards |
| 13:22.20 | *** part/#asterisk mythicalbox (n=mitchel@adsl-67-167-10.hsv.bellsouth.net) |
| 13:28.29 | Nilzao | edited /etc/asterisk/sip.conf created one extension |
| 13:28.59 | [TK]D-Fender | [09:19]<Nilzao>edited /etc/asterisk/dahdi_cfg <- pardon? |
| 13:29.18 | Nilzao | just jornaling, want me to stop? |
| 13:29.19 | [TK]D-Fender | Nilzao: that is not a valid config file name |
| 13:29.50 | Nilzao | oops |
| 13:29.54 | Nilzao | /etc/asterisk/chan_dahdi.conf |
| 13:29.56 | Nilzao | sorry |
| 13:29.59 | [TK]D-Fender | Nilzao: Go start a blog or before you know it people will "jornal" their tips the the BATHROOM here.... |
| 13:30.09 | [TK]D-Fender | trips* |
| 13:30.17 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 13:30.31 | Nilzao | bathroom is out of context |
| 13:31.42 | *** join/#asterisk theron (n=theron@216.51.246.211) |
| 13:31.46 | Nilzao | i will stop, just trying to show that i'm trying to make it by myself and not here to take free answers |
| 13:32.11 | *** join/#asterisk jeeger (n=user@192.68.211.161) |
| 13:32.14 | Nilzao | and i stopped to use the gui things, followed your advice |
| 13:35.33 | *** join/#asterisk j_kroon (n=jkroon@linux.delter.co.za) |
| 13:35.41 | [TK]D-Fender | Nilzao: Thas good to hear, but if everyony mentioed everything they do just fine with * it would be a continuous life-blog... |
| 13:35.49 | [TK]D-Fender | everyone* |
| 13:36.13 | Nilzao | indeed |
| 13:36.14 | Nilzao | sorry |
| 13:36.21 | Nilzao | but too quiet here too... |
| 13:36.23 | *** join/#asterisk GeekBoy (n=kvirc@200.94.227.117) |
| 13:36.39 | j_kroon | hi guys, i'n just wondering about outbound calls to DAHDI/ channels. My previous notion was that whenever asterisk dialed the number it "answered" the channel and bridged, however, now I'm getting NO ANSWER cdr records on calls that definitely passed out. |
| 13:36.44 | *** part/#asterisk GeekBoy (n=kvirc@200.94.227.117) |
| 13:36.52 | Nilzao | let's see how far i can get =] |
| 13:37.13 | j_kroon | I was hoping someone might be able to shed some light. |
| 13:37.41 | *** join/#asterisk ctaloi (n=Adium@nat-66-218-1-29.usadatanet.com) |
| 13:37.53 | beek | [TK]D-Fender: Write a bot to pull out the bathroom trip journals and then make it a website. I have no doubt there would be those willing to pay for access... |
| 13:38.36 | Nilzao | beek: if have photos and videos, they pay for access |
| 13:39.21 | beek | Nilzao: You have personal knowledge of this? ;-) |
| 13:39.23 | *** join/#asterisk jeeger (n=user@192.68.211.161) |
| 13:40.06 | [TK]D-Fender | j_kroon: depends on the channel's capacity to track call progress |
| 13:40.09 | Nilzao | beek: yes, people love to take care of other ppl life |
| 13:40.23 | coppice | if the toilet journal misses out urination and hand washing trips it will be like any other blog - 100% crap |
| 13:40.35 | j_kroon | [TK]D-Fender, analog, TDM800. |
| 13:40.46 | j_kroon | tone detection is about as reliable as non-existend. |
| 13:40.55 | [TK]D-Fender | j_kroon: Typically analog PSTN channels do not enable call progress and are indeed considered "answwered" once dialed |
| 13:41.18 | j_kroon | ok, so what would cause those calls to not get to ANSWERED but end up with NO ANSWER? |
| 13:41.54 | [TK]D-Fender | coppice: And stories about gas stations charging for use of their compressors to fill you tires is a sure sign of inflation. |
| 13:42.00 | j_kroon | client is complaining about beeping on some calls and the only correlation I can get is that when calls goes out over DAHDI/* and the log says NO ANSWER then it happens, otherwise it's fine. |
| 13:42.31 | [TK]D-Fender | j_kroon: call rpgress is something you can enable on Zaptel, but risks random disconnects, etc |
| 13:43.11 | j_kroon | thus why it states callprogress=no in chan_dahdi.conf. |
| 13:44.35 | jeeger | Greetings! I am trying to implement an incident callout with asterisk, and I fail to even make asterisk call me and play back the tt-weasels file (which sounds interesting^^). I create a call file and move it to the asterisk outgoing directory. It contains this: http://pastebin.com/f1f72fdcd . However, asterisk immediately hangs up on me instead of playing the sound, and in the log, I get the message "sent into invalid extension 's' in |
| 13:44.36 | jeeger | context 'default'. But I pass the context and extension in the call file, do I not? |
| 13:45.11 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:45.21 | *** join/#asterisk jtodd (n=Adium@c-76-124-123-117.hsd1.pa.comcast.net) |
| 13:45.21 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 13:45.33 | [TK]D-Fender | jeeger: You pas it AN extension, "s" if often used as a fall-back if what you ask for is not actually there |
| 13:45.55 | jeeger | Ah. Okay. |
| 13:45.59 | [TK]D-Fender | jeeger: Oh, and don't assume we take your word for it that it is ;) |
| 13:46.17 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.138.206) |
| 13:46.31 | jeeger | [TK]D-Fender: Yes, I'm looking right now^^ |
| 13:49.09 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:49.50 | jeeger | Yah, had to do a reload |
| 13:50.33 | jeeger | Ah, that worked. |
| 13:51.33 | *** join/#asterisk saftsack (n=saftsack@p57924321.dip.t-dialin.net) |
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| 14:00.30 | *** join/#asterisk Blues1976 (n=fortega@adsl-072-148-151-106.sip.mia.bellsouth.net) |
| 14:03.08 | Blues1976 | Hi. I installed AsteriskNow. I'm reading the "Asterisk The Future of Telephony book" . I'm stock at one point. in the CLI I type dialplan reload but it doesn't work... maybe I need a .conf file |
| 14:04.21 | Blues1976 | I think I'm missing extensions.conf |
| 14:04.29 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:04.50 | *** join/#asterisk Defraz (n=T0tal@63.228.246.229) |
| 14:05.18 | Blues1976 | but I notice that now, because that is in the analog part. the book said to skip that part if I was only doing IP. I'm assuming that I don't need the dialplan if I'm only going to use IP, right? |
| 14:07.41 | beek | Blues1976: You need a dialplan regardless of the technology you're using. |
| 14:09.48 | Blues1976 | beek: Yeah, I was begging to think that. for some reason that file is not under \etc\asterisk (extensions.conf) which is the file that I think is used for dial plan. there must be a template somewhere. |
| 14:09.53 | [TK]D-Fender | Blues1976: All calls are processed in the dialplan |
| 14:10.40 | [TK]D-Fender | Blues1976: there was a template.. when you installed... I'm not sure where there may be a backup in your install. If not, just DL the * tarball and extract the sample config |
| 14:11.06 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 14:12.09 | Blues1976 | D-Fender: Download asterisk-1.4.22.tar.gz and extract it from there... is that what you meant? |
| 14:12.13 | Blues1976 | thank you for the help |
| 14:12.15 | jeeger | Hm, I want to use Authenticate() with the j option, but it doesn't jump on error and instead hangs up the channel.... |
| 14:13.10 | beek | Blues1976: If you compiled from source did you dir a 'make samples' to get the initial configurations? |
| 14:13.33 | Blues1976 | beek: I installed AsteriskNow |
| 14:13.39 | beek | Oh. |
| 14:13.55 | jeeger | Okay, never mind that. |
| 14:14.32 | beek | Blues1976: In the download you'll find a configs directory. The files all end in '.sample', so look for 'extensions.conf.sample' |
| 14:14.52 | Blues1976 | Thanks for all the help. |
| 14:14.58 | Blues1976 | I will come back later! bye |
| 14:19.00 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:19.53 | [TK]D-Fender | jeeger: module unload res_falsealarm.so ;) |
| 14:21.24 | jeeger | But now another question: Is it possible to not call out at all? I want to create a dialplan that dials it's own numbers, and I don't want the call at the beginning. |
| 14:22.19 | [TK]D-Fender | jeeger: your dialplan can do whatever you tell it to |
| 14:22.52 | jeeger | But when I create an outgoing call with a .call file, I need to give a channel. Or can I just pass something bogus and jump into the dialplan? |
| 14:23.04 | *** join/#asterisk macros73 (n=cs_@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 14:23.22 | [TK]D-Fender | jeeger: And there is no such thing as "its numbers". Every number dialed by anything talking to * gets processed by your dialplan and be completely unique as compared to how any other call from any other device may be treated |
| 14:23.34 | [TK]D-Fender | jeeger: What is your goal? |
| 14:23.55 | [TK]D-Fender | jeeger: there has to be something on BOTH sides of the call. |
| 14:24.05 | jeeger | I want to call several people one after another. |
| 14:24.12 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:24.14 | [TK]D-Fender | jeeger: and do what with them? |
| 14:24.19 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:24.19 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:24.29 | jeeger | Ask them for a password and execute a script if it's correct. |
| 14:24.46 | [TK]D-Fender | jeeger: well you pointed them to that extension "1" in your dialplan. What is does is up to YOU. |
| 14:24.52 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:24.52 | *** join/#asterisk simprix (n=simprix@69.50.82.130) |
| 14:24.52 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:25.04 | [TK]D-Fender | jeeger: Noone said you have to dial out some other resource anywhere along the way |
| 14:25.37 | *** join/#asterisk Gopaul (n=Miranda@59.97.121.63) |
| 14:26.18 | jeeger | [TK]D-Fender: Problem is, these people don't call /me/, I call /them/. And when creating an outgoing call via a call file, I pass a channel name, and it seems that asterisk calls that channel immediately, without executing the dialplan, and only after the call is started, the dialplan gets control of the channel. |
| 14:27.02 | rhassing_work | I have some problems with my SIP peers... If the dnssrv is changed I get al lot of warnings (http://pastebin.com/d52da1e40) |
| 14:27.14 | rhassing_work | More people having these kind of problems? |
| 14:28.08 | [TK]D-Fender | jeeger: Whas is dialplan supposed to with without someone to IINTerACT with it? |
| 14:29.46 | jeeger | Call people and ask them for their passwords. |
| 14:30.16 | [TK]D-Fender | jeeger: thats what the "Channel;" line is for. Call out, when they answer, THEN dump them into the dialplan to process the call |
| 14:30.42 | jeeger | And if i wanted to call several people after another? Would I have to create several call files? |
| 14:31.28 | [TK]D-Fender | jeeger: Would you want to call multiple people regardless? |
| 14:31.43 | [TK]D-Fender | jeeger: Or stop after successfully getting ONE of them? |
| 14:32.07 | jeeger | the latter. |
| 14:32.24 | jeeger | and one after another, not at the same time. |
| 14:32.45 | jeeger | I thought I could just call Dial() in the dialplan, but that doesn't seem to work. |
| 14:33.08 | [TK]D-Fender | jeeger: that would be FORMER then. So you want call a bunch of people for the same purpose, not jsut try 1,2,3,4,5 until just one of them goes through. |
| 14:33.36 | [TK]D-Fender | jeeger: At which point this calls for 5 call files |
| 14:34.00 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:34.00 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:34.02 | jeeger | No, I want to call several people until I reach the first of them. |
| 14:34.10 | [TK]D-Fender | jeeger: And not call the others? |
| 14:34.20 | jeeger | No, I don't want to continue calling. |
| 14:34.41 | [TK]D-Fender | jeeger: Ok, clearer. |
| 14:35.33 | [TK]D-Fender | jeeger: then what you'll probably want to do is for your Channel: dial a LOCAL channel, and not a direct device. This way you can use the dialplan to do the dialing, check the status and on no answer continue on to dial the next |
| 14:35.45 | jeeger | Ah. I'll go look that up. |
| 14:35.45 | jeeger | thanks. |
| 14:36.13 | [TK]D-Fender | jeeger: chan_local the most important resource people never seem to get around to understanding. |
| 14:38.13 | *** join/#asterisk dni (n=dniz0r@74.169.15.252) |
| 14:38.43 | dni | hello all,. Can someone take a look at these little snippets of ym config and tell me what im doing wrong regarding setting callerid ,.. http://pastebin.com/m13d17b9f |
| 14:38.43 | *** part/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 14:42.26 | [TK]D-Fender | dni: What is the PROBLEM? |
| 14:42.35 | dni | It doesnt set the caller id |
| 14:42.40 | dni | ive also tried to define it in sip.conf |
| 14:42.54 | [TK]D-Fender | dni: normally you set "num" not "ani" |
| 14:42.55 | dni | like: callerid="Konstantinos Spyropoulos" <1001> |
| 14:43.11 | [TK]D-Fender | dni: And that also depends on what your provider allows |
| 14:43.42 | dni | My provider lets me set wwhatever i want on the caller id |
| 14:44.05 | [TK]D-Fender | dni: then use "num", not "ani" |
| 14:44.11 | dni | ok trying now |
| 14:44.17 | dni | there we go |
| 14:44.18 | *** join/#asterisk vvuja (n=vvuja@79-175-71-114.adsl-a-1.sezampro.yu) |
| 14:44.18 | dni | thanks |
| 14:44.21 | *** part/#asterisk vvuja (n=vvuja@79-175-71-114.adsl-a-1.sezampro.yu) |
| 14:44.43 | [TK]D-Fender | dni: and in your peer make sure you set "sendrpid=yes" |
| 14:49.02 | *** join/#asterisk deadpigeon (n=deadpige@office.xpressamerica.net) |
| 14:49.09 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-203-152.lns10.mel6.internode.on.net) |
| 14:55.06 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 14:55.48 | *** join/#asterisk propellerhead (n=yogurt2u@host133.190-30-26.telecom.net.ar) |
| 14:56.15 | Katty | pouts |
| 14:56.25 | Katty | anyone want to sell me a pink quilt |
| 14:57.08 | *** join/#asterisk jtodd1 (n=Adium@c-76-124-123-117.hsd1.pa.comcast.net) |
| 14:57.08 | *** mode/#asterisk [+o jtodd1] by ChanServ |
| 14:57.14 | deadpigeon | ... no. |
| 14:57.32 | Katty | :< |
| 14:57.43 | deadpigeon | ive been on hold for 1 hour and 30 minutes with adtran so far. >.< |
| 14:57.59 | deadpigeon | Sorry Katty, sometimes life has got to suck. |
| 14:58.03 | MaliutaLap | Katty: didn't you get the memo?? black is the new pink! ;) |
| 14:58.11 | timeshell_atwork | ewww |
| 14:58.15 | Katty | black and pink quilt would be acceptable |
| 14:58.25 | Katty | tho i think i'd prefer pink and brown. |
| 14:58.28 | timeshell_atwork | goes shopping for a pink quilt |
| 14:58.30 | MaliutaLap | I only have black |
| 14:58.42 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 14:58.49 | MaliutaLap | and I'n not going to sell it ... although you _could_ share it with me ;) |
| 14:59.05 | Katty | ummm no. |
| 14:59.09 | Katty | boys smell. |
| 14:59.18 | MaliutaLap | but I have female DNA :P |
| 14:59.27 | deadpigeon | im wondering if i could get better tech support for adtran's products somewhere on irc. heh. |
| 14:59.46 | jameswf | Katty: only if their doinitrite |
| 14:59.56 | Katty | jameswf: huh? |
| 15:00.00 | jameswf | nm |
| 15:00.03 | Katty | look! i just want a pink quilt! |
| 15:00.15 | Katty | no male required! |
| 15:00.41 | jameswf | get one of those snuggie deals |
| 15:00.43 | MaliutaLap | Katty: but if we can't mail it to you ... ;P |
| 15:01.04 | Katty | you can deliver it, if you like. |
| 15:01.16 | jameswf | It is like 110 degrees here so anything looking like a quilt will be destroyed |
| 15:01.39 | MaliutaLap | Katty: sure ... you paying my air fare? |
| 15:02.01 | MaliutaLap | long way over the pond, not sure I could wal it |
| 15:02.09 | MaliutaLap | s/wal/walk/ |
| 15:02.38 | Katty | no, i just want the quilt |
| 15:02.49 | Katty | i have no use for you. |
| 15:02.56 | Katty | (BURN!) |
| 15:03.07 | jameswf | such violence... |
| 15:03.23 | timeshell_atwork | Katty, I like girls. :D |
| 15:04.11 | timeshell_atwork | Why is light red called pink, but light blue is just still called blue and light yellow is still called yellow? |
| 15:04.18 | rob0 | Depends how they're cooked. |
| 15:04.48 | Katty | scowls |
| 15:04.56 | coppice | light blue is called cyan, and light yellow is called cream |
| 15:05.06 | timeshell_atwork | No |
| 15:05.09 | timeshell_atwork | cyan isn't really blue. |
| 15:05.16 | timeshell_atwork | and cream isn't really yellow |
| 15:05.37 | jeeger | Argh, I'm trying to use the local channel to jump into a dialplan without calling anyone, but when I use Dial(), the call gets bridged with the local channel and the dialplan gets executed again. |
| 15:05.53 | coppice | pink isn't really light red either |
| 15:06.04 | timeshell_atwork | coppice yes it is. |
| 15:06.12 | timeshell_atwork | pink is the result of red + white. |
| 15:06.30 | jameswf | jeeger: whats wrong with goto |
| 15:06.36 | timeshell_atwork | just as light blue is the result of blue + white |
| 15:06.42 | coppice | spectrally cream is the same as this, but peaking in the yellow area |
| 15:06.43 | jameswf | pink is salmon |
| 15:06.45 | Chainsaw | Please don't confuse additive & subtractive colour mixing. |
| 15:06.52 | SuPrSluG | azure |
| 15:06.56 | [TK]D-Fender | jeeger: you need to clearly look at what is on the side being CALLED "Channel: " and when it is answering. then whoever that is actually connected to, look at where you are dumping them in your dialplan. |
| 15:06.59 | timeshell_atwork | salmon is not pink. Salmon has orange in it |
| 15:07.19 | timeshell_atwork | white isn't a color |
| 15:07.27 | coppice | ducks have orange in them. salmon is grilled with lemon |
| 15:07.27 | jameswf | white is all color |
| 15:07.33 | MaliutaLap | Katty: well, in _that_ case you can come and get it :P |
| 15:07.34 | [TK]D-Fender | FUSCHIA SCREEN OF DEATH! |
| 15:07.41 | timeshell_atwork | YAH!! |
| 15:07.43 | jameswf | mmmm duck with orange... |
| 15:07.53 | jameswf | TB Green.... |
| 15:07.53 | timeshell_atwork | [TK]D-Fender 1 Spectrum 0 |
| 15:07.55 | jameswf | mmmm |
| 15:08.06 | MaliutaLap | [TK]D-Fender: they need a gayer colour than that |
| 15:08.10 | [TK]D-Fender | jameswf: Agent Orange? |
| 15:08.13 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 15:08.15 | rob0 | Rabbit season! |
| 15:08.23 | MaliutaLap | duck season ... shoot! |
| 15:08.32 | [TK]D-Fender | MaliutaLap: Lavender should do, or mauve, or "flseh" |
| 15:08.34 | timeshell_atwork | And how about infra-yellow? |
| 15:08.36 | [TK]D-Fender | flesh* |
| 15:08.43 | Katty | MaliutaLap: yeah, no... |
| 15:08.44 | SuPrSluG | is this the stream of asterisk channel |
| 15:09.01 | jameswf | You can hunt any animal year round you simply need to yell "THERE COMING RIGHT FOR US" |
| 15:09.15 | MaliutaLap | [TK]D-Fender: I was thinking something pale green, with a really gay name |
| 15:09.24 | Katty | seafoam |
| 15:09.31 | eppigy | hi |
| 15:09.32 | [TK]D-Fender | SuPrSluG: Yes, important to know the difference between "stream of consciousness" and "Field & Stream" ;) |
| 15:09.35 | Katty | hi dave |
| 15:09.37 | eppigy | hi |
| 15:09.39 | timeshell_atwork | Katty : UGH |
| 15:09.40 | Katty | i want to buy a pink quilt |
| 15:09.42 | [TK]D-Fender | Katty: EW... should have dies with the 60's |
| 15:09.43 | eppigy | :D |
| 15:09.44 | Katty | do you have one you want to sell? |
| 15:09.44 | jameswf | I have always been bothered by Mauve |
| 15:09.50 | MaliutaLap | Katty: shush ... you don't want my quilt :P |
| 15:09.52 | eppigy | I only have one blanket |
| 15:09.56 | eppigy | and one pair of sheets |
| 15:09.59 | eppigy | I am a simple man |
| 15:10.00 | Katty | :< |
| 15:10.02 | Katty | k |
| 15:10.14 | timeshell_atwork | Peuce! |
| 15:10.15 | SuPrSluG | no you are dave |
| 15:10.15 | [TK]D-Fender | Katty: I had a milkshake mizer in sea-foam green.... looked like a throwback, but I loved it... |
| 15:10.17 | Katty | eppigy: MaliutaLap is annoying me. |
| 15:10.25 | MaliutaLap | says something about Katty and pink bits |
| 15:10.37 | eppigy | MaliutaLap: wow how dare you |
| 15:10.44 | eppigy | HOW DARE YOU |
| 15:11.11 | Katty | i think you have something better to talk about than Katty and pink bits. |
| 15:11.14 | jeeger | [TK]D-Fender: http://pastebin.com/f2ad4c4e6 . Here's the dialplan and the call file. |
| 15:11.14 | MaliutaLap | eppigy: it's easy ... I press keys on this think called a "keyboard" |
| 15:11.27 | Katty | eppigy: all i wanted was a pink quilt. |
| 15:11.27 | eppigy | That is quite a feat when you can't read |
| 15:11.31 | eppigy | congratulations |
| 15:11.34 | rhassing_work | Katty, http://www.snugaustraliauggboots.com.au/index.php?currency=EUR&cPath=45&gclid=CKfVtYKXy5oCFcE63god8kVS4A |
| 15:11.39 | Katty | eppigy: i'm going to start pretending to be a male in my mid 40s. |
| 15:11.43 | eppigy | yesh |
| 15:11.50 | rob0 | Dare goes dat wascally wabbit! |
| 15:11.54 | eppigy | I enjoy There Will Be Blood |
| 15:11.56 | [TK]D-Fender | jeeger: You seem to completely MIX the "left" and "right" side concept in that call-file and exten |
| 15:12.29 | jeeger | Yeah. I noticed. However, I am somewhat confused about what constitutes the "left" and "right" side in a local channel context. |
| 15:12.32 | [TK]D-Fender | jeeger: One side should ONLY dial, because that reaches the person. the OTHER should do the actually actions once answers. you do not jsut follow a Dial command with call processing |
| 15:12.53 | Katty | rhassing_work: not quite the kind of quilt i had in mind. |
| 15:12.54 | eppigy | Hello I am Daniel Plainview |
| 15:12.57 | Katty | rhassing_work: here, have a picture: |
| 15:13.03 | eppigy | this is my son JT |
| 15:13.07 | *** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-8a5085860f3a8c52) |
| 15:13.07 | [TK]D-Fender | jeeger: "channel:" is who you call. who you call should not have authenticate in that exten |
| 15:13.50 | jeeger | Shoot, I have a Channel and Context: and Extension: in my call file. |
| 15:13.58 | jeeger | that explains why I get called twice. |
| 15:14.00 | Katty | rhassing_work: http://www.blossomquiltworks.com/images/FirstQuiltPinkQuilt.jpg |
| 15:14.01 | [TK]D-Fender | jeeger: and I clearly see you made a "callinc' and aren't USING it. Looks like you did half the job and forgot why you were doing it |
| 15:14.08 | [TK]D-Fender | jeeger: yup |
| 15:14.20 | jeeger | um, Macro(callinc,...) doesn't call the macro? |
| 15:14.23 | timeshell_atwork | jeeger : Doesn't your authenticate have the wrong goto on faiL? |
| 15:14.25 | [TK]D-Fender | jeeger: get that head screwed on straight, you aren't too far off... |
| 15:14.29 | jeeger | ^^ |
| 15:14.31 | jeeger | Thanks. |
| 15:14.51 | *** join/#asterisk Aiatek (n=Asterisk@75.112.88.200.m.sta.codetel.net.do) |
| 15:15.03 | [TK]D-Fender | jeeger: Umm.. actually, that is a mess, I'm feeling generous, gimme a sec |
| 15:15.24 | Katty | http://mamabearcreations.com/yahoo_site_admin/assets/images/quilt_004.171201734_std.jpg <- that would also be very cute. |
| 15:15.44 | rhassing_work | Katty, And this one: http://www.overstock.com/Home-Garden/Porter-Quilt-Set/2589732/product.html :-) |
| 15:16.12 | Katty | hmm. pink. |
| 15:16.15 | Katty | bit too plain tho. |
| 15:16.26 | rhassing_work | But it is REALLY pink ;) |
| 15:16.32 | Katty | it is! |
| 15:16.34 | timeshell_atwork | Katty : http://cgi.ebay.ca/40-5-QUILT-Sqs-CHUTES-LADDERS-MODA-PINKS-BROWN-BL_W0QQitemZ400050113095QQcmdZViewItemQQptZUS_Fabric?hash=item400050113095&_trksid=p3286.c0.m14&_trkparms=72%3A1215|66%3A2|65%3A12|39%3A1|240%3A1318|301%3A1|293%3A1|294%3A50 |
| 15:16.42 | Katty | also not a quilt )= |
| 15:16.57 | Katty | a quilt is a bunch of smaller squares of fabric sewn together with somethign in the middle. |
| 15:16.59 | jake[work] | i'm confused - was there a workaround completed for the sonus dtmf issues? |
| 15:17.26 | Katty | timeshell_atwork: also not a quilt :< |
| 15:17.55 | timeshell_atwork | http://cgi.ebay.ca/GIRLS-HOT-PINK-BUTTERFLY-DOUBLE-QUILT-COVER-DUVET-SET_W0QQitemZ230342928582QQcmdZViewItemQQptZUK_Home_Garden_Bedroom_Bedding_PP?hash=item230342928582&_trksid=p3286.c0.m14&_trkparms=72%3A1215|66%3A2|65%3A12|39%3A1|240%3A1318|301%3A0|293%3A1|294%3A50 |
| 15:17.58 | Katty | i might have to just make one |
| 15:18.22 | Katty | pretty! |
| 15:18.24 | Katty | but again, not a quilt )= |
| 15:18.26 | timeshell_atwork | Here one for MaliutaLap: http://cgi.ebay.ca/BLACK-CERISE-PINK-DOUBLE-QUILT-COVER-DUVET-SET_W0QQitemZ280344090146QQcmdZViewItemQQptZUK_Home_Garden_Bedroom_Bedding_PP?hash=item280344090146&_trksid=p3286.c0.m14&_trkparms=72%3A1215|66%3A2|65%3A12|39%3A1|240%3A1318|301%3A0|293%3A1|294%3A50 |
| 15:18.38 | [TK]D-Fender | jeeger: http://pastebin.com/m2737c329 |
| 15:18.44 | eppigy | D: |
| 15:18.58 | eppigy | NEIN |
| 15:19.07 | Katty | http://www.whi.org/quilts/quilt.jpg <- this is a quilt |
| 15:19.19 | Katty | block of material sewn together to form a larger blanket. |
| 15:19.19 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 15:19.27 | Katty | i want a base pink, with brown blocks. |
| 15:19.47 | eppigy | oh nice |
| 15:20.00 | jake[work] | do u guys know what the irc channel is for asterisk? |
| 15:20.10 | eppigy | this one |
| 15:20.12 | Katty | http://thechaly.files.wordpress.com/2007/08/quilt-1.jpg <- another sample of a quilt. |
| 15:20.13 | eppigy | sometimes |
| 15:20.29 | eppigy | jake[work]: my heart is filled with anger |
| 15:20.32 | eppigy | how are you? |
| 15:20.33 | jake[work] | :) |
| 15:20.46 | Katty | jake[work]: disolve that anger into some sugar and water! |
| 15:20.53 | jake[work] | i should prob ask on the dev side |
| 15:21.38 | rhassing_work | Just another 10 minutes and then it's weekend!!! |
| 15:22.40 | coppice | Katty: what brought up the patchwork quilt topic? did someone mention Windows Vista? |
| 15:23.11 | Katty | coppice: ummm. |
| 15:23.13 | Katty | coppice: i want one? |
| 15:23.40 | [TK]D-Fender | Whats all this talk about narcissism, nd what does that have to do with ME? |
| 15:25.25 | *** join/#asterisk CunningPike (n=CunningP@vpn.dnv.org) |
| 15:26.16 | MaliutaLap | timeshell_atwork: that's funny |
| 15:27.30 | timeshell_atwork | MaliutaLap : I knew you'd like it ;) |
| 15:28.23 | jeeger | [TK]D-Fender: Many thanks. I've understood the concept. But slowly it seems as if I better take the AGI script route. This dialplan would become very complex. |
| 15:28.50 | [TK]D-Fender | jeeger: Well I don't know how far you intend to take things. It'll be for you to decide |
| 15:29.04 | [TK]D-Fender | jeeger: But glad the concept is solid now. |
| 15:31.07 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 15:31.14 | jameswf | ok anyone off toppic gets B& |
| 15:31.45 | *** part/#asterisk rhassing_work (n=rob_work@ti152.telin.nl) |
| 15:33.33 | *** join/#asterisk propellerhead (n=yogurt2u@host180.190-225-213.telecom.net.ar) |
| 15:33.44 | Aiatek | i want to take the Dcap exam and when i tried to contact the trainning center they told me 'we only have seats for those who takes the trainning' |
| 15:35.13 | jameswf | Aiatek: /msg jsmith he is the training main dude he may have a good answer or solution |
| 15:35.25 | *** join/#asterisk mnicholson_ (n=mnichols@nat/digium/x-69fbb79d77d3e124) |
| 15:35.42 | Aiatek | where he is? |
| 15:36.15 | jameswf | Aiatek: probably in /dev but you do not need to be in the same room to message him |
| 15:36.22 | jameswf | ~seen jsmith |
| 15:36.25 | infobot | jsmith <n=njsmith@asterisk/training-and-documentation-guru/jsmith> was last seen on IRC in channel #utah, 18h 18m 16s ago, saying: 'goozbach: Yes, it is better...'. |
| 15:36.53 | jaytee | #utah? |
| 15:37.00 | Katty | shivers. |
| 15:37.09 | Aiatek | ok |
| 15:37.33 | coppice | I guess "Yes, it is better" wasn't about the channel's topic |
| 15:37.46 | *** join/#asterisk stevedude77 (n=stevedud@63.68.135.4) |
| 15:38.10 | Aiatek | i think he is not online right now, i will be around |
| 15:38.11 | Aiatek | thx |
| 15:38.16 | rob0 | /j #mormon |
| 15:39.00 | Katty | eppigy: wanna go get some lunch |
| 15:39.22 | eppigy | yeah dude |
| 15:39.27 | eppigy | I am getting hungry |
| 15:39.30 | eppigy | D: |
| 15:39.31 | [TK]D-Fender | [11:33]<Aiatek>i want to take the Dcap exam and when i tried to contact the trainning center they told me 'we only have seats for those who takes the trainning' <- this does not sound like a legitimate practice whatsoever. I'd definitely take it up with him |
| 15:40.54 | Kobaz | i hate dtmf |
| 15:41.09 | jaytee | I hate smurfs |
| 15:41.49 | Katty | eppigy: what should we havce |
| 15:42.24 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 15:42.31 | Kobaz | i have voicepulse on one box, voip.ms on another.... i can dial out voicepulse and dtmf goes through.... i can dial from my landline and the dtmf goes to the voip.ms box.... but if i dial out voicepulse to the voip.ms box... there is NO DTMF |
| 15:42.38 | *** part/#asterisk stevedude77 (n=stevedud@63.68.135.4) |
| 15:43.54 | Kobaz | so dtmf gets lost somewhere between voicepulse and voip.ms |
| 15:44.21 | Kobaz | you would think, if i sent it inband... it would make it through... but that doesn't work |
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| 15:47.15 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:47.19 | eppigy | Katty: I am really craving a mango,avacado,tuna roll |
| 15:49.19 | Katty | eppigy: hmm. |
| 15:49.22 | Katty | eppigy: where do you get that? |
| 15:49.35 | Katty | avacado sounds dreamy. |
| 15:49.48 | *** join/#asterisk outtolunc (n=me@c-71-198-197-201.hsd1.ca.comcast.net) |
| 15:50.11 | Kobaz | [TK]D-Fender: so umm... hehe |
| 15:50.16 | Katty | or maybe a nice tuna melt. |
| 15:50.17 | Kobaz | [TK]D-Fender: any ideas? :) |
| 15:50.19 | Katty | that also sounds dreamy. |
| 15:50.51 | eppigy | its is at this sushi buffet called nori nori |
| 15:58.44 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
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| 16:00.31 | [TK]D-Fender | eppigy: yum. i've got a decent place here that we for lunch for $15 AYCE |
| 16:00.47 | [TK]D-Fender | eppigy: Dinner is $25 |
| 16:00.56 | *** join/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 16:01.03 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 16:01.17 | jplank | anyone in the uk right now? specifically london? |
| 16:01.58 | coppice | probably 8 to 10 million people should be in london right now |
| 16:01.59 | eppigy | yesh |
| 16:02.07 | eppigy | my place is very similar |
| 16:02.21 | jameswf | <aplause> |
| 16:02.38 | jplank | ok, in this channel specifically |
| 16:03.21 | jplank | I just want to make sure I have the timezone right |
| 16:03.24 | *** join/#asterisk seste (n=rseste@mail.daitanlabs.com) |
| 16:03.38 | [TK]D-Fender | jplank: GMT 0 |
| 16:03.58 | [TK]D-Fender | jplank: Doesn't take a native to answer that |
| 16:04.02 | jplank | thats what I have, and daylight savings time would push it to BST? |
| 16:04.45 | [TK]D-Fender | jplank: Stop artificially limiting you answer-base |
| 16:05.09 | coppice | yes, Bull Shit Time is GMT+1 |
| 16:05.19 | jplank | yea, people where busting my ass last night when asking the question, because I was asking too late for people from london |
| 16:05.24 | jplank | busting my balls* |
| 16:06.29 | [TK]D-Fender | jplank: You know this kind fo question falls squarely in JFGI territory, and you should probably be dragged out and shot for even bothering to ask... right? ;) |
| 16:06.59 | jplank | I knew the answer, but I just wanted to make sure |
| 16:07.17 | [TK]D-Fender | jplank: 1/2 a milion hits can't be wrong ;) |
| 16:07.19 | jplank | and having the customer calling me telling me the time was wrong was not how I wanted to confirm :) |
| 16:07.45 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 16:08.11 | [TK]D-Fender | heads out to lunch |
| 16:08.17 | [TK]D-Fender | jplank: You get of easy.. THIS time. |
| 16:08.27 | jplank | lol |
| 16:08.28 | jplank | thanks |
| 16:08.51 | jplank | just wait, I have a install in china next month |
| 16:08.54 | jplank | ;) |
| 16:09.22 | coppice | CST is GMT+8, and doesn't change in summer |
| 16:09.41 | jplank | yea, I'm just kidding, the time zone is easy there, the dial map is a different story |
| 16:09.52 | jplank | might just do the good old 9|. |
| 16:12.21 | coppice | the dial map in China is pretty straightforward. they just make the description hard to follow |
| 16:13.21 | coppice | and they screw up lots of services by answering the phone to play you "he's not around" announcements |
| 16:14.58 | *** join/#asterisk Nilzao (n=nils@200-168-146-103.dsl.telesp.net.br) |
| 16:15.13 | *** part/#asterisk jake[work] (n=Jake@pool-173-52-144-183.nycmny.east.verizon.net) |
| 16:17.41 | jarg | what server do you recomend for this board: 1AEX2406EF - 24 port modular analog PCI-Express x1 card with 24 Trunk interfaces and HW Echo |
| 16:17.57 | Dovid | TK: snt a PM |
| 16:17.58 | jarg | i'm worry about the size |
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| 16:22.44 | *** join/#asterisk ddickenson (n=ddickens@rrcs-97-77-245-251.sw.biz.rr.com) |
| 16:24.04 | ddickenson | is there a way to send a ring group to a single voicemail box and light indicators on several phones but not the entire ring group |
| 16:25.53 | *** join/#asterisk SebastianS (n=schu@adsl-dyn215.78-98-80.t-com.sk) |
| 16:29.40 | bmoraca | uhg...polycom's sales certification "course" is so rife with bullshit buzzwords it's making me sick |
| 16:30.18 | *** join/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 16:30.43 | asteriskmonkey | anyone seen a bug in asterisk 1.6.1 where there is one way audio after being passed from an IVR? |
| 16:32.52 | eppigy | has anyone created a good asterisk reporting platform |
| 16:33.43 | bmoraca | the one that's built in to freepbx (which is also available standalone) works pretty well...i've liked using it |
| 16:33.59 | bmoraca | i can't remember what it's called, though |
| 16:40.05 | *** join/#asterisk cosmo83 (n=arava@117.195.165.71) |
| 16:41.29 | cosmo83 | Hi guys . Can someone guide me in a call recording appliance iam looking for. I have a simple FXO device . if i use a analog splitter and connect one end to a telephone and other end to an asterisk box. define the ZAP channels and outbound trunk, can i record the call ? |
| 16:41.56 | *** join/#asterisk spck (n=spck@unioncab.com) |
| 16:42.02 | spck | tada! |
| 16:43.29 | jameswf | well Qwell looks like *Now is making our production line up... |
| 16:45.33 | *** join/#asterisk hi365 (n=hi365@94.159.178.51) |
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| 16:55.29 | *** join/#asterisk Bilbolodz (n=bilbo@pc-bilbo.man.lodz.pl) |
| 16:55.40 | Bilbolodz | hi all, |
| 16:56.18 | Bilbolodz | I desperatelly need some help with dahdi! Could some one can help me? |
| 16:56.46 | bmoraca | well, no one can really answer that question until you tell us what problems you're having |
| 16:57.59 | Bilbolodz | fresh install: dahdi 2.1.0.4 dahdi-tools 2.1.0.4 asterisk 1.4.24.1 and fresh os (debian stable). |
| 16:58.13 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 16:58.31 | MaliutaLap | nudges tzafrir_laptop |
| 16:58.51 | Bilbolodz | dahdi is correctly loaded card is detected and configured but asterisk can't see any dahdi channels |
| 17:00.06 | Bilbolodz | what's going on??? |
| 17:00.36 | MaliutaLap | Bilbolodz: it has to do with the debian builds ... you're better off with zaptel and those packages |
| 17:01.13 | Bilbolodz | sorry I didn't get it. Could you repeat? |
| 17:01.25 | MaliutaLap | Bilbolodz: I just went to upgrade from there to 1.6.1 (on unstable) and there are some glaring holes in the builds |
| 17:01.47 | Bilbolodz | ok but is 1.4.24.1 |
| 17:01.48 | MaliutaLap | Bilbolodz: is a simple solution ... purge the dahdi packages and install the zaptel ones |
| 17:01.52 | tzafrir_laptop | MaliutaLap, what is it? |
| 17:01.56 | *** join/#asterisk ruben23 (n=AGENT@124.107.3.178) |
| 17:02.12 | Bilbolodz | ok I will try with zaptel |
| 17:02.23 | Bilbolodz | but is it know issue? |
| 17:02.30 | MaliutaLap | tzafrir_laptop: are you aware there are 2 dependancies that can't be satisfied in the 1.6.1 packgage :) |
| 17:02.56 | Bilbolodz | I'm compiling from sources |
| 17:03.01 | Bilbolodz | not deb's |
| 17:03.13 | tzafrir_laptop | Bilbolodz, asterisk? zaptel? dahdi? |
| 17:03.18 | Bilbolodz | all |
| 17:03.20 | Nilzao | as i know asterisk 1.4 uses zaptel |
| 17:03.27 | Nilzao | 1.6 uses dahdi |
| 17:03.41 | Bilbolodz | 1.4 can use dahdi too |
| 17:03.49 | Bilbolodz | I think |
| 17:03.49 | Nilzao | never tryed... |
| 17:03.58 | Bilbolodz | ok I will install zaptel |
| 17:04.47 | Nilzao | dudes... now my turn... |
| 17:04.56 | bmoraca | wow...the SoundStation IP 7000 multi-unit connectivity option is really, really expensive... |
| 17:05.05 | Nilzao | with asterisk-gui we have the users.conf |
| 17:05.13 | bmoraca | ~users.conf |
| 17:05.13 | infobot | users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 17:05.33 | Nilzao | there set the trunks |
| 17:05.42 | Nilzao | as users |
| 17:05.47 | Nilzao | then send to extensions... |
| 17:06.05 | Nilzao | without the users.conf how i can set the way? |
| 17:06.16 | Nilzao | have 2 fxo ports |
| 17:06.21 | Nilzao | one is working |
| 17:06.25 | bmoraca | sip.conf + extensions.conf = much better control over everything |
| 17:06.35 | Nilzao | ok, i saw that |
| 17:06.43 | Nilzao | that's why i'm reconfiguring all |
| 17:06.51 | Nilzao | reinstalled a new system |
| 17:07.05 | Nilzao | i was journaling this morning, but they said to stop |
| 17:07.24 | Nilzao | have to DAHDI |
| 17:07.31 | Nilzao | the DAHDI/1 and the DAHDI/2 |
| 17:07.53 | bmoraca | Nilzao: when you use a GUI, you have to be ready to accept the limitations of said GUI. if you need support for a particular GUI, I would check in the IRC room for that GUI. such as #asterisk-gui |
| 17:08.05 | Nilzao | i'm not at asterisk-gui |
| 17:08.17 | Nilzao | i reinstalled all without it |
| 17:08.28 | Nilzao | i'm comparing the way to make the thing |
| 17:08.39 | Nilzao | trying to make the 2 lines receive calls |
| 17:09.06 | Nilzao | just set at [default] |
| 17:09.06 | Nilzao | exten=>s,Answer |
| 17:09.07 | Nilzao | exten=>s,Dial(SIP/username) |
| 17:09.21 | Nilzao | the DAHDI/2 works |
| 17:09.34 | Nilzao | the DAHDI/1 don ring at the CLI |
| 17:10.35 | bmoraca | what's your zapata.conf look like? |
| 17:10.53 | bmoraca | ~pb |
| 17:10.54 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 17:10.56 | Nilzao | i'm using the dahdi.conf |
| 17:11.06 | bmoraca | right. what's it look like? |
| 17:11.08 | Nilzao | rebooting asterisk, i pb soon |
| 17:11.32 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 17:12.00 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 17:13.48 | *** join/#asterisk reallost1 (n=reallost@c-75-66-45-46.hsd1.tn.comcast.net) |
| 17:14.37 | Nilzao | http://pastebin.com/d406062fd |
| 17:14.49 | Nilzao | when i plug the line at the board |
| 17:15.03 | Nilzao | the asterisk CLI say somethings about RED alarm |
| 17:16.15 | Nilzao | detected alamr on channel 1: Red Alarm |
| 17:16.18 | Nilzao | *alrm |
| 17:16.20 | Nilzao | *alarm |
| 17:16.21 | ruben23 | hi guys. |
| 17:16.26 | reallost1 | Hey |
| 17:16.29 | Nilzao | sup ruben |
| 17:17.00 | *** join/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc) |
| 17:17.28 | ruben23 | got problem registering my remote sip client to my public asterisk server |
| 17:17.39 | mercutioviz | quick TDM question: has anyone here ever used OSLEC with a Sangoma analog card, like the A400? |
| 17:17.39 | Nilzao | what the CLI says about it? |
| 17:17.53 | ruben23 | no firewall, no Selinux all disbaled |
| 17:18.04 | Nilzao | not behind a router? |
| 17:18.16 | ruben23 | not behind router.. |
| 17:18.22 | Nilzao | and the CLI says? |
| 17:18.27 | ruben23 | my asterisk uses public Ip |
| 17:18.42 | ruben23 | i got request time out |
| 17:18.48 | ruben23 | on my softphones |
| 17:19.03 | Nilzao | the account have nat enabled? |
| 17:19.08 | Nilzao | nat=yes |
| 17:19.30 | ruben23 | no nat yes... |
| 17:19.39 | ruben23 | do i need to put it..? |
| 17:19.48 | [TK]D-Fender | ruben23: .... |
| 17:19.50 | Nilzao | try, don forget to restart asterisk |
| 17:19.51 | [TK]D-Fender | ~sipnat |
| 17:19.52 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:19.54 | [TK]D-Fender | ^^^^^ |
| 17:20.07 | [TK]D-Fender | ruben23: How many times do we have to link to yout he guide? |
| 17:20.36 | ruben23 | [TK]D-Fender: hi... |
| 17:21.15 | ruben23 | infobot:my asterisk server is not behind NAt..it uses a public IP |
| 17:21.16 | infobot | ruben23: okay |
| 17:21.17 | *** join/#asterisk s0lid (n=s0lid@122.53.108.176) |
| 17:21.39 | Nilzao | lol |
| 17:21.39 | asteriskmonkey | i have a really odd issue, if i call a user direct audio works fine both ways, yet when calling through an ivr on the samebox we have 1 way audio... tried asterisk 1.6.1 and astersik 1.6.2-branches |
| 17:21.44 | Nilzao | talking with infobot |
| 17:22.07 | Nilzao | ruben23: just try to enable the nat=yes at your sip conf |
| 17:22.27 | Nilzao | ruben23: its free =] |
| 17:22.29 | spck | infobot: i would like a sandwich with corned beef and swiss on rye with thousand island dressing, please |
| 17:22.30 | infobot | You would like a sandwich with corned beef and swiss on rye with thousand island dressing, please? |
| 17:22.41 | ruben23 | Nilzao: ok ill do that:) |
| 17:22.59 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 17:23.18 | spck | infobot: a moose once bit my sister |
| 17:23.33 | *** join/#asterisk nullable_type (n=nullable@hq.verbx.net) |
| 17:23.38 | spck | ^^ |
| 17:23.43 | Nilzao | well one of my DAHDI not ringing in the CLI |
| 17:23.46 | lizor | laughs |
| 17:23.59 | _ShrikE | wik |
| 17:24.02 | Nilzao | sent the pb |
| 17:24.47 | rob0 | The producers wish to apologise. Those responsible for the credits have been sacked. |
| 17:25.39 | nullable_type | Hey guys, do the calling card companies use Asterisk? Does Asterisk scale as the concurreny calls goes up (let's say 1000) with a dedicated system |
| 17:31.33 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:34.17 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 17:34.19 | ruben23 | quick question here: as of last day when i check my CLI ouput i go color display on its log now, when i try to view it i only see white text no color display on logs. anything unusual..? |
| 17:35.54 | Nilzao | ruben23: verbose mode lower? |
| 17:36.09 | Nilzao | ruben23: or modified the logger.conf |
| 17:37.24 | spck | is it possible to cluster in such a way that you won't drop active calls if a server fails? |
| 17:38.11 | *** join/#asterisk mykhyggz (n=mykhyggz@evolone.org) |
| 17:39.03 | Nilzao | spck: i guess if the call using an analog line is not possible |
| 17:39.33 | Nilzao | spck: like you calling from the analog line at server1, then it halts... the server2 can't help you |
| 17:41.07 | Aiatek | <spck> i think thats not possible |
| 17:41.16 | spck | eh wishful thinking |
| 17:41.56 | spck | even with a drbd implementation? |
| 17:43.22 | Aiatek | nope |
| 17:43.35 | Aiatek | drbd wont keep the actives call |
| 17:43.58 | Aiatek | you make a failover to the second node |
| 17:44.07 | Nilzao | spck: you can make a server to hold all the hardware, and 2 servers with cluster |
| 17:44.30 | nullable_type | Hey guys, do the calling card companies use Asterisk? Does Asterisk scale as the concurreny calls goes up (let's say 1000) with a dedicated system |
| 17:44.33 | Aiatek | when you make the failover the second server will start all the related services |
| 17:44.44 | Aiatek | like asterisk, mysql, etc |
| 17:46.08 | Aiatek | you can use redfone for digital hardware |
| 17:49.02 | ruben23 | Nilzao:this is the output of my logger.conf http://pastebin.com/m6c49d50d |
| 17:49.22 | *** join/#asterisk ManxPower (n=manxpowe@245.sub-70-214-85.myvzw.com) |
| 17:49.39 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 17:50.18 | *** join/#asterisk lasko_ (n=lasko@70.102.15.210) |
| 17:50.33 | [TK]D-Fender | spck: Depends on which piece of equipement fails, what is distributing calls, what role * is filling, what you are using for termination, etc |
| 17:50.33 | *** part/#asterisk lasko_ (n=lasko@70.102.15.210) |
| 17:50.52 | [TK]D-Fender | Aiatek: And RedFone is a flaming piece of shit |
| 17:51.17 | [TK]D-Fender | Aiatek: I sorry, that is imprecise... |
| 17:51.37 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 17:51.37 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 17:51.38 | Bilbolodz | HELP!!!! I've installed new system, and compiled fresh zaptel and fresh astersik. The same situation: *CLI> zap show channels |
| 17:51.38 | Bilbolodz | <PROTECTED> |
| 17:51.57 | Nilzao | did u modprobe the zaptel? |
| 17:51.57 | Aiatek | <[TK]D-Fender> its that bad? |
| 17:52.02 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-37a107024b7e21c9) |
| 17:52.05 | [TK]D-Fender | Aiatek: RedFone is a fresh steamy pile of shit in a braises based of crap with sprinkles on top. |
| 17:52.07 | *** join/#asterisk javb (n=javb@tdev212-102.codetel.net.do) |
| 17:52.09 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 17:52.17 | [TK]D-Fender | braised* |
| 17:52.36 | javb | had someone connected before the USB - FXO Sangoma Adapter? I cant see it installing Zaptel 1.4. . . |
| 17:52.50 | Bilbolodz | yes zaptel is running cat: /proc/zaptel/: Is a directory |
| 17:52.50 | Bilbolodz | debian:~# cat /proc/zaptel/1 |
| 17:52.50 | Bilbolodz | Span 1: WCTDM/0 "Wildcard S400P Prototype Board 1" (MASTER) |
| 17:52.50 | Bilbolodz | <PROTECTED> |
| 17:52.50 | Bilbolodz | <PROTECTED> |
| 17:52.50 | Bilbolodz | <PROTECTED> |
| 17:52.59 | [TK]D-Fender | Bilbolodz: PASTEBIN, do not spam in here |
| 17:53.00 | [TK]D-Fender | ~pb |
| 17:53.01 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 17:53.03 | [TK]D-Fender | ^^^^^^^6 |
| 17:53.10 | Bilbolodz | ok sorry |
| 17:53.27 | Aiatek | <[TK]D-Fender> what problems can i have with redfone? |
| 17:53.39 | Nilzao | ~redfone |
| 17:53.45 | Nilzao | ~redphone |
| 17:53.56 | Nilzao | infobot: wtf is redfone? |
| 17:53.58 | Aiatek | i havent see it in production |
| 17:54.06 | [TK]D-Fender | Aiatek: The only thing it talks to is *, iffy hardware & support. |
| 17:54.25 | Nilzao | infobot: wtf <redfone> |
| 17:54.34 | Nilzao | i give up |
| 17:54.36 | Nilzao | lets google it |
| 17:54.41 | [TK]D-Fender | Nilzao: JFGI |
| 17:54.59 | Nilzao | ~jfgi |
| 17:55.00 | infobot | http://www.google.com/search?q=jfgi |
| 17:55.08 | Nilzao | cool |
| 17:55.17 | Aiatek | dosent work well with asterisk? |
| 17:55.17 | Bilbolodz | so? Any ideas whats wrong with my zap channels |
| 17:55.18 | Bilbolodz | ? |
| 17:56.50 | Nilzao | so you /etc/zaptel.conf is ok, and your /etc/asterisk/zaptel.conf? |
| 17:56.50 | [TK]D-Fender | Aiatek: it depends on * so don't expect ti to survive * crashing and not losing calls |
| 17:57.18 | Aiatek | i dont said you wont lose a call |
| 17:57.29 | Aiatek | i know that because i know about HA |
| 17:57.38 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 17:57.51 | [TK]D-Fender | Aiatek: Yes, well its trouble to configure, poor support, no ineroperability. Lose-lose |
| 17:57.54 | Aiatek | i just wanna know if the redfone its a so bad or not |
| 17:57.59 | Aiatek | ok |
| 17:58.02 | [TK]D-Fender | Aiatek: So don't get stuck with crap |
| 17:58.07 | Bilbolodz | I think yes: http://pastebin.ca/1428986 |
| 17:58.09 | Aiatek | thats what i was asking you |
| 17:58.23 | Aiatek | technical details |
| 17:58.29 | Aiatek | not "its a shit" |
| 17:58.32 | [TK]D-Fender | Aiatek: RedFone is a fresh steamy pile of shit in a braised based of crap with sprinkles on top. <- I really thought this would be clear :) |
| 17:58.46 | [TK]D-Fender | Aiatek: yeah, not a specific, but a fair warning just the same |
| 17:59.32 | [TK]D-Fender | Bilbolodz: Your zapata.conf has no [channels] header ad KILLS the config. it will not load because of that |
| 17:59.51 | [TK]D-Fender | Bilbolodz: And you are missing a lot of other things you should probably be setting |
| 18:00.07 | Superbartt | Dear friends ( :P ) I was asked to build an asterisk system with 4 E1/T1 cards, being able to handle about 100~120 concurrent calls... What hardware do you guys recommend? Just a plain Xeon Quadcore or multiple of them? |
| 18:00.18 | n3hxs | Aiatek look at their web page, advertising AstriCon 2008... attention to detail? I think not. |
| 18:01.02 | Nilzao | Superbartt: buy the Xeon Quadcore, then keep an eye on the processor |
| 18:01.16 | Nilzao | Sueprbartt: if need more, you put more processors =] |
| 18:01.25 | Bilbolodz | ok where I will find sample zapata.conf? |
| 18:01.41 | Nilzao | Bilbolodz, just put [channel] at start |
| 18:01.41 | Superbartt | Nilzao also an idea |
| 18:01.45 | [TK]D-Fender | Bilbolodz: In the source tarball. |
| 18:01.59 | [TK]D-Fender | Superbartt: "with 4 E1/T1 cards" <- you mean 4 PORTS, right? |
| 18:02.13 | [TK]D-Fender | Superbartt: And the rest depends on what you are doing with your calls |
| 18:02.44 | Superbartt | [TK]D-Fender yes that's what i meant :p |
| 18:03.34 | Bilbolodz | finally! |
| 18:03.36 | Bilbolodz | thaks |
| 18:03.49 | Superbartt | connect them 1 one 1 to a SIP phone [TK]D-Fender :p |
| 18:03.52 | Bilbolodz | correct is: [channels] |
| 18:04.25 | [TK]D-Fender | Superbartt: Local? What codec? Call recording? |
| 18:05.12 | Superbartt | Yes, uhmm interally probally alaw, and some random recording |
| 18:05.42 | ruben23 | Nilzao:.. |
| 18:05.45 | [TK]D-Fender | SuPuthen yeah a single multi-core zeon with 4 GIG should be plenty comfortable |
| 18:05.57 | Nilzao | sup ruben23 |
| 18:06.26 | ruben23 | Nilzao:this is the output of my logger.conf http://pastebin.com/m6c49d50d |
| 18:06.31 | Qwell | [TK]D-Fender: Zeon? knockoff? |
| 18:07.50 | ruben23 | Bilbolodz: is it working now..? |
| 18:07.50 | Superbartt | [TK]D-Fender Is it ok if I'll just use plain xeon's? Zeon sounds like Zoltan from dude where's my car ;) |
| 18:07.50 | [TK]D-Fender | Qwell: ok ok , I slipped :p |
| 18:07.50 | Nilzao | ruben23: it's ok, what you need on it? |
| 18:08.16 | ruben23 | Nilzao: still got plain white text output logs on my asterisk CLI |
| 18:08.30 | ruben23 | no color hinting |
| 18:08.50 | [TK]D-Fender | Superbartt: O RLY? Quoting from "Dude, Where's My Car?" Doesn't exactly make YOU look bright :) |
| 18:08.55 | Nilzao | ruben23: you mean like warning red notice green? |
| 18:08.55 | ruben23 | what could be its cause.,just last night its fine |
| 18:08.59 | *** join/#asterisk tonyplee (i=41db0407@gateway/web/ajax/mibbit.com/x-a96c4d695370c291) |
| 18:09.00 | ruben23 | yes |
| 18:09.03 | ruben23 | thats it |
| 18:09.06 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 18:09.10 | Superbartt | lol [TK]D-Fender =) |
| 18:09.17 | ruben23 | now its a plain white text logs |
| 18:09.47 | Nilzao | ruben23: are you using the same terminal to access the ssh? |
| 18:10.15 | ruben23 | im using extraputty now form my client PC |
| 18:10.50 | Nilzao | ruben23: do you see colors like blue dirs or green executables when ls? |
| 18:11.19 | ruben23 | Nilzao: yes i can see colors |
| 18:11.33 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 18:11.40 | Nilzao | ruben23: hold on, i'm jfgi |
| 18:12.48 | ruben23 | Nilzao:ok |
| 18:14.23 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 18:14.43 | Nilzao | take a look /etc/asterisk/asterisk.conf if the nocolors=no |
| 18:15.01 | *** join/#asterisk ddickenson (n=ddickens@rrcs-97-77-245-251.sw.biz.rr.com) |
| 18:15.34 | ddickenson | anybody know how to set a ring group to point a a single mailbox and get the indicator to light on 3 of the 7 phones in the group? |
| 18:17.32 | [TK]D-Fender | ddickenson: "ring group" is a vague term, and your Dial has nothing to do with what VM box you choose to let a call fall to |
| 18:17.49 | n3hxs | So you want the mailbox to light 3 phones when a message is left? ddickenson? |
| 18:18.10 | ddickenson | yes |
| 18:19.00 | ddickenson | I have dial app dialing seven phones when a certain number comes in from outside but they want a single mailbox that they all fall to and light mw lamp on 3 of the phones... |
| 18:19.02 | n3hxs | doesn't have the answer but I wanted to clarify the question in my mind. |
| 18:19.47 | ddickenson | it's a "main line" to this clinic |
| 18:21.02 | *** join/#asterisk iksik (i=xk@livedata.pl) |
| 18:21.07 | iksik | hi |
| 18:21.44 | iksik | i've got this problem only with one of the users: Registration from '"6680"<sip:adi@domain.com;transport=UDP>' failed for 'IP.HERE' - No matching peer found |
| 18:21.45 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1279581612.dsl.bell.ca) |
| 18:21.49 | ddickenson | http://pastebin.com/m5871157 |
| 18:21.50 | iksik | any ideas how to fix it ? |
| 18:23.18 | Nilzao | try type=friend at sip.conf |
| 18:23.27 | Nilzao | iksik: try type=friend at sip.conf |
| 18:23.40 | iksik | it's set |
| 18:23.45 | [TK]D-Fender | ddiddthen put the mailbox line in each devices config |
| 18:24.27 | *** join/#asterisk ctp (n=ctp@brsg-d9befcb8.pool.mediaWays.net) |
| 18:24.28 | [TK]D-Fender | ddickenson: then put the mailbox line in each device's config |
| 18:25.06 | rhassing | ddickenson, in sip.conf you can configure the same mailbox for those 3 phones |
| 18:25.09 | ddickenson | what if I used a macro to create the extensions? do I need to just make those 7 on their own lines so I can add voicemail |
| 18:25.18 | ddickenson | ohhh |
| 18:25.36 | [TK]D-Fender | ddickenson: device config, not dialplan. |
| 18:25.41 | iksik | Nilzao, any other ideas? :( |
| 18:26.02 | rhassing | ~pb |
| 18:26.03 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 18:26.09 | ddickenson | I was looking in extensions.conf. so if the lines are 1701, 1702, 1703... etc and the ring group is 3535 I just set mailbox for those 3 extensions to 3535 |
| 18:26.11 | *** part/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc) |
| 18:26.21 | rhassing | iksik, pate your config, so we can have a look |
| 18:26.30 | rhassing | s/pate/paste |
| 18:26.44 | Nilzao | iksik: sorry bro, i'm kinda * newbie, when it happened the type=friend worked for me |
| 18:27.02 | rhassing | ddickenson, yep :) simple isn't it :) |
| 18:27.09 | iksik | http://pastebin.com/m66c3c4ed |
| 18:27.20 | iksik | Nilzao, i'm noobie to ;P |
| 18:27.31 | ddickenson | seems that way, you don't happen to know the syntax off hand do you? |
| 18:28.39 | rhassing | mailbox=3535@default |
| 18:28.56 | ddickenson | oh, I probably should have just guessed that! |
| 18:29.04 | rhassing | :) |
| 18:29.23 | ddickenson | thanks |
| 18:29.37 | rhassing | ddickenson, np, you're welcome |
| 18:30.50 | *** join/#asterisk propellerhead (n=yogurt2u@host44.190-136-118.telecom.net.ar) |
| 18:32.54 | rhassing | iksik, It looks ok... Can you ping the phone (the default ip address which is set)? |
| 18:33.40 | *** join/#asterisk cyford (n=allen@12.22.184.2) |
| 18:37.47 | cyford | can some one tell me why my asterisk is rejecting this number from my trunk , i get this message. [May 15 13:21:06] NOTICE[24357] chan_sip.c: Call from '' to extension '9678791XXXX' rejected because extension not found. |
| 18:37.54 | iksik | rhassing, hm, checking |
| 18:37.56 | Nilzao | oh boys... you know the old school analog pbx? |
| 18:38.14 | Nilzao | that the ring makes "ring ring", not "riiiiiing" |
| 18:38.41 | iksik | rhassing, nope, I can't ping it :| |
| 18:39.06 | Nilzao | the generic clone board cant receive the "ring ring" |
| 18:39.17 | iksik | rhassing, this user is behind NAT I think. |
| 18:39.18 | rhassing | iksik, can you check the ip address on this phone to see if it was set correctly |
| 18:39.21 | Nilzao | that was why not ringing... |
| 18:39.32 | Nilzao | cheap hardware sux |
| 18:40.06 | iksik | Internal IP + some STUN thing :| |
| 18:41.57 | rhassing | iksik, add nat=yes to the sip.conf for this user and canreinvite=no |
| 18:42.24 | iksik | ok |
| 18:42.40 | rhassing | iksik, but htat doesn't help you to register, for that you should not set defaultip |
| 18:42.51 | iksik | uhm |
| 18:42.51 | iksik | ok |
| 18:43.10 | rhassing | host=dynamic will give * the ip address of this user |
| 18:43.32 | iksik | hmm |
| 18:43.37 | iksik | still same error |
| 18:43.40 | iksik | Received SIP subscribe for peer without mailbox: (null) |
| 18:43.42 | iksik | and this one |
| 18:43.43 | iksik | :| |
| 18:45.34 | rhassing | iksik, can you paste the error to pastebin? |
| 18:46.37 | iksik | rhassing http://pastebin.com/m1f585706 |
| 18:46.44 | rhassing | cyford, is 9678791XXXX in your dialplan? "show dialplan 9678791XXXX@<context> |
| 18:47.00 | *** join/#asterisk MrTelephone (n=test@h697179-171.picriverisp.net) |
| 18:47.16 | MrTelephone | it looks like sip spamming or brute force attacks are getting more common |
| 18:47.23 | MrTelephone | does anyone notice this? |
| 18:48.00 | rhassing | iksik, did you delete the line with defaultip from the sip.conf? |
| 18:48.17 | iksik | yes |
| 18:49.39 | rhassing | iksik, could it be a firewall issue? |
| 18:49.51 | iksik | mine firewall? |
| 18:50.26 | iksik | mine account works fine... |
| 18:50.52 | rhassing | or on the other side. to register a sip client you would need to open port 5060 |
| 18:51.17 | iksik | and this one is werid for me: <sip:adi@domain_here.com;transport=UDP> |
| 18:51.27 | rhassing | and if you would like to speak to each other you need to open the rtp ports as well |
| 18:51.38 | iksik | when I'm connecting into server I have not noticed that TRANSPORT=UDP |
| 18:52.39 | rhassing | iksik, can you paste the current sip.conf for this user? |
| 18:53.17 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:53.20 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 18:53.35 | iksik | http://pastebin.com/m1dc54830 |
| 18:56.27 | rhassing | iksik, looks ok to me... and the first package gets to the server, so the phone is able to reach the *. Maybe the way back is a problem. I mean for * to reach this phone |
| 18:57.42 | iksik | mabe I've mess with extensions.conf ? I don't understand it yet... / |
| 18:58.01 | MrTelephone | what does it usually mean when the phone rings back when you hit the "END" button on a cordless handset? |
| 18:58.12 | rhassing | iksik, extensions.conf is not involved in the registration of a client |
| 18:58.18 | iksik | uhm |
| 18:59.43 | iksik | rhassing Can I PM with You? I can paste logs from softvoip of this user |
| 18:59.52 | iksik | mabe that could help |
| 18:59.55 | rhassing | iksik, ok |
| 19:00.12 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 19:06.20 | seste | hi all |
| 19:06.42 | seste | I'm having some problems with chan_mobile |
| 19:07.28 | seste | I can hear no sound on both directions... |
| 19:07.54 | seste | I'm using asterisk and asterisk addons 1.6.1.0 |
| 19:08.03 | seste | and bluez 4.32 |
| 19:08.41 | atraxxx | Can anyone recommend a good place to post an Asterisk Admin job? Tried cragislist, with only a few hits, and DICE charges $600! Any specific mailing list for this type of thing, or should I just bite the bullet at put it on DICE and/or Monster? |
| 19:09.03 | *** join/#asterisk lasko (n=lasko@70.102.15.210) |
| 19:09.10 | *** part/#asterisk lasko (n=lasko@70.102.15.210) |
| 19:09.32 | SuPrSluG | try voip-info |
| 19:11.04 | seste | I enable the debug from asterisk and saw that the function mbl_right/read is not called. |
| 19:11.13 | rhassing | atraxxx, connect to LinkedIn (www.linkedin.com) |
| 19:11.14 | seste | could some one help me?? |
| 19:11.37 | Katty | no |
| 19:12.21 | rhassing | seste, I dont know function mbl_right |
| 19:12.25 | carrar | YES WE CAN |
| 19:12.27 | ruben23 | Nilzao:sorry just got back form a break.. |
| 19:12.43 | Nilzao | ruben23: don worry, i was fighting with my * here |
| 19:13.06 | ruben23 | i laready look at it i got this inputs http://pastebin.com/m5e6baae |
| 19:13.29 | seste | rhassing this fucntion writes data in the audio socket |
| 19:16.02 | seste | sorry....it is mbl_write |
| 19:16.03 | Nilzao | ruben23: what file is this |
| 19:16.39 | ruben23 | asterisk.conf |
| 19:16.50 | Nilzao | put + 2 lines |
| 19:17.10 | Nilzao | [options] |
| 19:17.10 | Nilzao | nocolor=no |
| 19:17.50 | atraxxx | rhassing: thanks.. just noticed there's a JOBS tab on the asterisk users group in LinkedIn. No Jobs have been posted there yet.. I wonder why. |
| 19:20.30 | ruben23 | [Nilzao:thi is my output......is this rigth......? http://pastebin.com/m55dce681 |
| 19:20.57 | Nilzao | ruben23: yes, worked? |
| 19:21.36 | ruben23 | ill try it now |
| 19:22.16 | ruben23 | Nilzao: hmmm..it did not work.. |
| 19:22.23 | ruben23 | i make reload |
| 19:22.23 | MaliutaLap | oh yay. chan_dadhi won't load and I can only find 2 refernces to the issue with google, one is not in english, one is a channel log for here with an unanswered question |
| 19:23.14 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 19:23.37 | *** join/#asterisk af_ (n=getsmart@88-149-240-107.dynamic.ngi.it) |
| 19:23.40 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 19:24.24 | Nilzao | ruben23: sorry man, i don't know what happening |
| 19:25.13 | ruben23 | Nilzao:i understand |
| 19:25.30 | jameswf | Citel phone would be alot better with a quality manual... |
| 19:26.49 | *** part/#asterisk Bilbolodz (n=bilbo@pc-bilbo.man.lodz.pl) |
| 19:27.13 | MaliutaLap | right. which one of you wants to help with a chan_dahdi loading issue? |
| 19:27.15 | MaliutaLap | http://pastebin.com/m375cd2f7 |
| 19:28.07 | jameswf | MaliutaLap: Perhaps a log file with more verbosity |
| 19:30.05 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 19:31.29 | MaliutaLap | jameswf: even with -vvvv thats all the info I get |
| 19:31.34 | *** join/#asterisk bgmarete (n=marebri_@196.201.210.130) |
| 19:32.17 | Nilzao | MaliutaLap: you can put level 5 verbose, and edit the logger.conf to verbose more |
| 19:33.48 | Nilzao | what version of asterisk? |
| 19:34.29 | *** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130) |
| 19:34.45 | *** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130) |
| 19:35.52 | MaliutaLap | Nilzao: 1.6.1 |
| 19:36.47 | *** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130) |
| 19:39.52 | SuPrSluG | have you rebuilt dahdi and asterisk? |
| 19:39.58 | Nilzao | MaliutaLap: you compiled this chan_dahdi.so? |
| 19:40.06 | MaliutaLap | and it doesn't seem to matter what I do with verbosity levels and logger.conf, it doesn't want to give up any more info |
| 19:40.23 | MaliutaLap | no, this is from the debian-voip teams packages |
| 19:40.35 | MaliutaLap | looks like I may have to build from scratch |
| 19:40.41 | SuPrSluG | yes |
| 19:40.58 | MaliutaLap | was hoping to avoid that, the only i386 I have is the * box and it is slow as hell |
| 19:40.59 | Nilzao | MaliutaLap: check the kernel-version this binary was build |
| 19:41.00 | SuPrSluG | always build from source |
| 19:41.26 | MaliutaLap | SuPrSluG: not _always_, I know when to do that and when not to |
| 19:41.31 | *** join/#asterisk nullable_type (n=nullable@hq.verbx.net) |
| 19:41.48 | Nilzao | MaliutaLap: not to do when have a 386? |
| 19:41.59 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 19:42.01 | Nilzao | MaliutaLap: and do when have a Xeon quad? =] |
| 19:42.24 | MaliutaLap | Nilzao: it's a celeron coppermine 733, it runs i386 binaries |
| 19:42.33 | nullable_type | Hey guys, I installed g729 from digium but somehow when i try to make a call using g729 codec, i get no audio formats found errror. Can someone help? |
| 19:42.47 | MaliutaLap | nullable_type: do you have a license? |
| 19:42.48 | Nilzao | MaliutaLap: just check the kernel version the debian-voip team made the binary |
| 19:43.11 | nullable_type | MaliutaLap >> Yes I do, i registered it too and i have the lincence in the correct folder |
| 19:43.20 | nullable_type | *paid licence |
| 19:43.27 | MaliutaLap | Nilzao: shouldn't make a difference, I built the dahdi kernel modules myself |
| 19:43.58 | Nilzao | you said: no, this is from the debian-voip teams packages |
| 19:44.00 | MaliutaLap | nullable_type: what does 'module show like g729' show? |
| 19:44.41 | Nilzao | now what? you built it or not? |
| 19:44.42 | nullable_type | it says 2 module loaded format_g729.so |
| 19:44.48 | MaliutaLap | Nilzao: the * binary is, as is the chan_dahdi.so ...the modules are built for this kernel. and the rest of dahdi is setup fine |
| 19:44.50 | nullable_type | * 1 module loaded |
| 19:44.55 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 19:45.07 | nullable_type | I am using asterisk 1.6.1 |
| 19:45.18 | KyleK | is there anything I can add to this dial command so it takes 20 seconds before going to the end line in extensions.conf if the two phones are unavailable? Dial(SIP/line1&SIP/xlite,20) |
| 19:45.20 | MaliutaLap | nullable_type: and 'g729 show licenses' |
| 19:45.33 | KyleK | s/end/next/ |
| 19:45.53 | nullable_type | MaliutaLap >> No such command is what i get |
| 19:46.31 | MaliutaLap | nullable_type: did you put codec_g729.so in the right place? |
| 19:46.41 | MaliutaLap | nullable_type: because the codec isn't loaded |
| 19:46.54 | MaliutaLap | nullable_type: load codec_g729a.so |
| 19:47.20 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 19:47.20 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 19:47.32 | nullable_type | yes i put in the modules folder but the codec is called codec_g729a.so though not codec_g729.so |
| 19:47.48 | nullable_type | do i need to rename it to remove the letter a? |
| 19:47.53 | MaliutaLap | nullable_type: load codec_g729a.so |
| 19:47.55 | MaliutaLap | no |
| 19:48.06 | MaliutaLap | it's called me being lazy |
| 19:50.11 | Katty | bored. |
| 19:50.46 | nullable_type | MaliutaLab >> I get a message saying load failed |
| 19:50.50 | MaliutaLap | Katty: too bad, you already rejected me once tonight :P |
| 19:51.06 | Katty | i have no need for male company. |
| 19:51.10 | MaliutaLap | nullable_type: did you get the right version of the module for your cpu type? |
| 19:51.10 | Katty | i already have a male that supplies that. |
| 19:51.44 | MaliutaLap | Katty: oh, because I'm really a candidate for "company" being on the other side of the world |
| 19:51.55 | Katty | precisely. |
| 19:52.00 | Deeewayne | Katty, do you need a pet squirrel ? |
| 19:52.04 | nullable_type | yes, i had the source downloaded and placed when another developer installed in with 1.4 Asterisk, I didn't re-download though, i used same |
| 19:52.11 | nullable_type | do i need to redownload? |
| 19:52.12 | Katty | Deeewayne: i have 5 pets, i think i'm good. |
| 19:52.22 | MaliutaLap | nullable_type: you can't get source for the codec |
| 19:52.28 | nullable_type | sorry i mean the binary |
| 19:52.48 | MaliutaLap | nullable_type: it souds like you don't have the right codec file ... you are aware you can't use the 1.4 file |
| 19:52.51 | nullable_type | was downloaded and installed in an older version of Asterisk, I just upgraded Asterisk and re-registered the codec |
| 19:53.08 | nullable_type | oh ya i c...... may be thats why i will check the knowledgebase |
| 19:53.13 | MaliutaLap | nullable_type: go to downloads.digium.com and get the appropriate 1.6 |
| 19:53.26 | nullable_type | Maliuta >> Will i need to re-register? Because it only allows to register cetain times |
| 19:53.41 | MaliutaLap | nullable_type: no, just drop the file in place and load it |
| 19:53.51 | nullable_type | alright great! Thank you so much |
| 19:54.10 | MaliutaLap | nullable_type: be sure you get the right one though ... I just had my 1.6.1 segfault because of the wrong one |
| 19:54.19 | Katty | i have this feeling....i think it's called aggitation. |
| 19:54.20 | Katty | irritation. |
| 19:54.23 | Katty | being annoyed? |
| 19:54.23 | nullable_type | oh ok alrite |
| 19:54.36 | *** join/#asterisk ACK-NAK (n=Miranda@home.chicagoventure.com) |
| 19:54.44 | MaliutaLap | Katty: go hit someone :P |
| 19:54.49 | nullable_type | MaliutaLap >> Are you a developer for Asterisk? |
| 19:54.58 | MaliutaLap | nullable_type: no |
| 19:55.20 | Katty | ah. google says disgruntled. |
| 19:55.47 | *** join/#asterisk ghenry (n=ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 19:55.54 | MaliutaLap | disgruntled is a different type of annoyed |
| 19:55.58 | MaliutaLap | more targeted |
| 19:56.35 | nullable_type | I am not an Asterisk or core C developer, but i made some modificaton to http module to find me if a call is in progress using an AccountCode easily. Should i submit it, Will it be useful for anyone? |
| 19:57.00 | Katty | "showing or experiencing dissatisfaction or restless longing" |
| 20:00.57 | Corydon76-dig | nullable_type: is it something that could be accomplished easily using either func_curl or res_config_curl ? |
| 20:01.52 | *** join/#asterisk ingenius (n=alektro@netsolution.com.ar) |
| 20:02.10 | n3hxs | is Gruntled ;) |
| 20:02.27 | nullable_type | Cory >> Actually its just an extra function that is exposed through HTTP manager api, gives you a list of curreny call with a given accountcode |
| 20:03.29 | nullable_type | actually i think there is an existing one, i just simplified to be able to easily parse |
| 20:04.18 | Corydon76-dig | nullable_type: if you use the XML output, it should be pretty easy to parse already |
| 20:05.11 | nullable_type | ya i guess, nevermind, it is just a simplified solution just for my requirement |
| 20:06.25 | Corydon76-dig | nullable_type: It's nice that you found it useful; but I'm more interested in patches that do what you cannot do already |
| 20:06.39 | eppigy | Katty: longing |
| 20:07.02 | *** join/#asterisk SebastianS_ (n=schu@adsl-dyn58.78-98-36.t-com.sk) |
| 20:08.06 | MaliutaLap | I have a longing, but she's not back in town until saturday :( |
| 20:08.45 | [TK]D-Fender | [15:45]<KyleK>is there anything I can add to this dial command so it takes 20 seconds before going to the end line in extensions.conf if the two phones are unavailable? Dial(SIP/line1&SIP/xlite,20) <- Yes, also dial a local channel that does Wait(20) |
| 20:09.42 | nullable_type | Cory >> I agree with you. BTW I find it very hard to get the canreinvite work, Did you ever have done direct RTP working? |
| 20:12.12 | nullable_type | ManliLap >> I was looking for g729 in digium downloads for Intel Xeon 4, I don't see any, Should i use Pentium4 codec? |
| 20:12.41 | KyleK | theres no benchmarker for g729? |
| 20:12.41 | ACK-NAK | Anybody know how to make voicemail Last-in-first-out instead of FIFO? Particularly for OLD messages? I didn't see an option for it in voicemail.conf. Is it a new feature of 1.6.0? 1.6.1? |
| 20:13.10 | MaliutaLap | nullable_type: are you running in 32bit mode? |
| 20:13.15 | nullable_type | yes |
| 20:13.36 | *** join/#asterisk juanIMP (n=Juancho@190.146.167.188) |
| 20:14.00 | nullable_type | http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.6.1/x86-32/ |
| 20:14.34 | nullable_type | My system: Intel Xeon 4 3.0 GHz |
| 20:15.00 | MaliutaLap | Nugget: probably the i686 note pentium comes in 3m 4m and m |
| 20:15.19 | MaliutaLap | s/Nugget/nullable_type/ |
| 20:15.25 | nullable_type | lol ok |
| 20:16.04 | MaliutaLap | I'm not up with what the intel codenames are these days |
| 20:16.14 | MaliutaLap | oh, and I couldn't give a rats :P |
| 20:16.36 | nullable_type | shouldn't i just use the i686 one that doesn't say the processor |
| 20:17.11 | MaliutaLap | that was my point |
| 20:17.25 | MaliutaLap | unless you know you have a prescott or something else |
| 20:17.38 | nullable_type | lol ok |
| 20:17.40 | nullable_type | thank you! |
| 20:20.54 | Nugget | eyes MaliutaLap |
| 20:21.21 | [TK]D-Fender | hands Nugget a spoon |
| 20:21.45 | MaliutaLap | Nugget: it's your own fault ... you shouldn't have a nick so close to someone else for tab completion :P |
| 20:25.24 | *** join/#asterisk SlipperyChicken (n=andrew@CPE0013f7c51659-CM0013f7c51655.cpe.net.cable.rogers.com) |
| 20:28.39 | Katty | BUT WHY A SPOON COUSIN |
| 20:29.03 | docid | because.... SPOON!!!! ...sorry, i really loved the tick |
| 20:29.18 | Katty | blinks. |
| 20:29.22 | Corydon76-dig | There is no spoon. |
| 20:29.22 | Katty | that did not parse. |
| 20:29.38 | [TK]D-Fender | BBIAB |
| 20:29.41 | Katty | i guess no one got the reference :< |
| 20:29.49 | MaliutaLap | Katty: I'm too busy cancelling christmas |
| 20:30.13 | MaliutaLap | Katty: because it's blunt you ... |
| 20:30.17 | voxter | anyone ever tried to get a polycom vtx1000 to speak to asterisk? |
| 20:30.45 | Katty | MaliutaLap: you're a mean one. |
| 20:30.56 | nullable_type | MaliutaLap >> It seemed the system i had was prescott and i downloaded the codec for it seems to work great |
| 20:30.57 | timeshell_atwork | voxter does it speak sip? |
| 20:31.02 | Katty | MaliutaLap: you as cuddly as a cactus. |
| 20:31.23 | Katty | MaliutaLap: you're a monster. |
| 20:31.24 | MaliutaLap | Katty: that depends on who's doing the cuddling |
| 20:31.33 | Katty | MaliutaLap: your brain is full of spiders. |
| 20:31.37 | timeshell_atwork | Cancelling christmas? |
| 20:31.59 | MaliutaLap | Katty: and yours is like a steel trap ... full of mice :P |
| 20:32.25 | MaliutaLap | timeshell_atwork: it's a reference to Robin Hood: Prince of Thieves ... keep up |
| 20:32.34 | Katty | MaliutaLap: stink. stank. stunk! |
| 20:32.56 | MaliutaLap | Katty: you have a skunk in your bed? |
| 20:33.11 | Katty | MaliutaLap: you nauseate me. |
| 20:33.29 | MaliutaLap | Katty: ahh, that's good to know |
| 20:33.46 | Katty | MaliutaLap: you're a 3 decker sour kraute and toadstill sandwich, with arsenic sauce! |
| 20:33.59 | MaliutaLap | mmm sour kraute |
| 20:34.15 | MaliutaLap | Katty: I prefer digitalis to arsenic, it's sweeter |
| 20:34.26 | Katty | compliments of Boris Karloff. |
| 20:34.41 | Katty | or perhaps, Thurl Ravenscroft. |
| 20:35.06 | Katty | REF: http://www.youtube.com/watch?v=MPBS7dVrE1U |
| 20:35.14 | MaliutaLap | Katty: you know "Maliuta" was the forerunner of Beria? and all that was evil in eastern europe? |
| 20:35.41 | MaliutaLap | I don't use this nick for fun |
| 20:38.13 | Katty | hungry :< |
| 20:38.43 | Katty | eppigy: :< |
| 20:39.06 | *** join/#asterisk cyford (n=allen@12.22.184.2) |
| 20:40.27 | eppigy | :< |
| 20:40.36 | eppigy | Katty: I am hungry as well |
| 20:40.38 | MaliutaLap | I should cook something for breakfast |
| 20:40.46 | MaliutaLap | I could have that roo that's in the fridge |
| 20:41.22 | n3hxs | Hop to it MaliutaLap ;) |
| 20:42.37 | javb | what is dadhi linux "complete" ? |
| 20:42.45 | Nilzao | who is 212.235.70.195 ? |
| 20:42.45 | timeshell_atwork | a pain in the neck |
| 20:43.20 | timeshell_atwork | javb In my opinion, it's better to download and install separately |
| 20:43.51 | javb | timeshell_atwork, which one do you install first? |
| 20:44.06 | timeshell_atwork | Don't remember. |
| 20:44.30 | timeshell_atwork | javb, But if you don't install the correct one first, I believe the other will complain |
| 20:45.00 | Katty | eppigy: corndog craving. |
| 20:45.12 | *** join/#asterisk hff135 (i=464017c7@gateway/web/ajax/mibbit.com/x-6ea317ac028499cc) |
| 20:45.21 | hff135 | hi all |
| 20:45.44 | MaliutaLap | hff135: no, we haven't smoked anything ... yet |
| 20:46.30 | hff135 | i have an audio problem. we recently set up some new asterisk boxes and audio quality is poor. dropped words for both sides of the conversation |
| 20:46.35 | KyleK | Nilzao: someone in Israel? |
| 20:46.50 | Nilzao | cool |
| 20:47.02 | *** join/#asterisk ks3 (n=ks3@74.203.195.1) |
| 20:47.04 | hff135 | i ran zttest and it came back with about 99.95%. this is lower than is recommended. i'm wondering if this is the cause of my problems |
| 20:47.09 | Nilzao | KyleK: nice, israel users tryin to register at my sip asterisk |
| 20:47.21 | rhassing | javb, first libpri (if needed), then dahdi, then dahdi-tools and then Asterisk |
| 20:47.21 | MaliutaLap | Nilzao: I think you want a console and "whois 212.235.70.195" |
| 20:47.42 | hff135 | please note that i don't have have any PSTN PRI's or lines plugged into the box |
| 20:47.43 | ks3 | Is there a way to specify which context gets checked for extensions on a 302 redirect? Right now any redirects are being searched for in default. |
| 20:47.48 | hff135 | any ideas? |
| 20:47.49 | Nilzao | MaliutaLap: irc console? |
| 20:48.01 | KyleK | Nilzao: /bin/bash console |
| 20:48.04 | MaliutaLap | Nilzao: a *nix console/shell |
| 20:48.14 | MaliutaLap | KyleK: no /bin/csh |
| 20:48.22 | Nilzao | lol |
| 20:48.26 | Nilzao | not logged |
| 20:48.26 | KyleK | doesn't matter |
| 20:48.39 | Nilzao | i guess its fring server |
| 20:48.40 | MaliutaLap | KyleK: how about /bin/false? |
| 20:49.02 | KyleK | is /bin/false a shell? |
| 20:49.23 | MaliutaLap | KyleK: on my systems it it |
| 20:49.37 | MaliutaLap | s/it$/is/ |
| 20:49.44 | Nilzao | well i will nmap |
| 20:50.03 | MaliutaLap | hah! infobot is bamboozled by regex |
| 20:50.11 | *** join/#asterisk haryv (n=lanny@S010600a0c93f6f7e.vs.shawcable.net) |
| 20:50.25 | KyleK | MaliutaLap: on most systems /bin/false is a binary that just does "return 0" |
| 20:51.17 | MaliutaLap | KyleK: you can register it in /etc/shells and set it to be the shell for all those accounts that don't need a real shel |
| 20:51.22 | nephfl | i switched the card and still having the same problem with channel 3 not dialing with a call file |
| 20:51.44 | MaliutaLap | KyleK: same goes for /bin/true |
| 20:52.29 | nephfl | is there some order that i need to put the modules in or voodoo magic i need to perform to get this to work? |
| 20:52.37 | KyleK | sure it fits in /etc/shells but that doesn't make it a shell |
| 20:53.05 | MaliutaLap | KyleK: depends, it's not a shell you can use for interface |
| 20:53.15 | MaliutaLap | KyleK: it is one you can use to deny people access |
| 20:53.16 | KyleK | MaliutaLap: setting peoples shell to /bin/false is to NOT give them a shell |
| 20:53.22 | eppigy | Katty: dude that sounds really good |
| 20:53.28 | MaliutaLap | KyleK: that is the point |
| 20:53.32 | KyleK | MaliutaLap: shell means a CLI, not a variable |
| 20:53.46 | KyleK | soo /bin/false is NOT a shell |
| 20:53.51 | MaliutaLap | KyleK: you don't get it |
| 20:53.56 | KyleK | you dont get it |
| 20:54.22 | MaliutaLap | I think I understand why you put /bin/false in /etc/shells and use it as a login shell for accounts |
| 20:54.30 | KyleK | MaliutaLap: you could set someones shell to X and get the same lack of being able to do anything |
| 20:54.33 | Nilzao | just type "man false" |
| 20:54.46 | KyleK | does that make Xorg a shell? :) |
| 20:55.26 | MaliutaLap | if you conf it that way |
| 20:55.31 | Katty | eppigy: yes. |
| 20:55.41 | Katty | eppigy: but i don't know where to get them. |
| 20:55.47 | Katty | eppigy: the fair won't be around for awhile |
| 20:55.55 | MaliutaLap | KyleK: I wouldn't recommend that one, but you could do it |
| 20:56.00 | Nilzao | nice, the ip is from amsterdam |
| 20:56.30 | KyleK | MaliutaLap: I prefer my definition on what a shell is, yours is to vague :) |
| 20:57.02 | MaliutaLap | KyleK: a "shell" is just a program spawned (generally by login) once a user has logged in ... if it gives them an interface then that's great ... it might also spwan them into something else, like a telnet/ssh proxy |
| 20:57.30 | MaliutaLap | KyleK: you definition re |
| 20:58.13 | MaliutaLap | KyleK: your definition restrict the use of the shell section of /etc/passwd to giving people access to your box ... that's not alway desirable |
| 20:58.19 | eppigy | Katty: flee market? |
| 20:58.23 | eppigy | flea |
| 20:58.28 | eppigy | flee lol |
| 20:58.30 | KyleK | not really |
| 20:58.34 | eppigy | d: |
| 20:58.40 | nephfl | can someone help me troubleshoot this tdm card? |
| 20:59.04 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 20:59.17 | KyleK | I'm saying that putting /bin/false into /etc/shells doesn't make it a "shell", I'm not saying I wouldn't set someones shell to be /bin/false, I'd set it at will, but I would not refer to /bin/false as a shell in any conversation |
| 20:59.36 | nephfl | it is a tdm410, and i can use a call file on channel 4 but not channel 3 |
| 21:00.14 | *** join/#asterisk Solver (n=robert@CPE0050fcc6a940-CM001cea35fd4e.cpe.net.cable.rogers.com) |
| 21:01.29 | MaliutaLap | pokes Solver |
| 21:01.36 | MaliutaLap | Solver: you can take it up with KyleK |
| 21:02.03 | Solver | I'm busy! :) |
| 21:02.28 | Solver | bottom line many binaries can be used as a shell. /bin/false has been used in this way for decades |
| 21:02.32 | [TK]D-Fender | nephfl: It should have no relationshipw ith a cll file so much as any call |
| 21:02.48 | MaliutaLap | Solver: at least the decades I have been doing *nix |
| 21:02.49 | nephfl | what do you mean? |
| 21:03.04 | [TK]D-Fender | nephfl: call files have nothing to do with trouble ports |
| 21:03.14 | Solver | I missed the earlier discussion but caught it 2nd hand :) |
| 21:03.14 | KyleK | Solver: I'm saying /bin/false being set as peoples shells to not let them do anything doesn't mean its a 'shell' |
| 21:03.22 | nephfl | if i use a call file with channel 4 it rings the phone plugged into the zap card if i do channel 3 it doesnt |
| 21:03.28 | [TK]D-Fender | nephfl: Defective is defective, its not due to being used by a call-file |
| 21:03.38 | nephfl | this is a new card |
| 21:03.41 | nephfl | new modules |
| 21:03.43 | Solver | KyleK: ah so it is a definition issue. It is not a shell which will let them do anything useful |
| 21:03.46 | nephfl | had it overnighted |
| 21:03.50 | Solver | which is the point I suppose :) |
| 21:03.52 | KyleK | hahaha |
| 21:04.02 | KyleK | Solver: thanks for agreeing with me :) |
| 21:04.09 | Solver | hahah |
| 21:04.15 | MaliutaLap | KyleK: actually he was agreeing with me :) |
| 21:04.27 | MaliutaLap | KyleK: /usr/bin/ircii is also a shell |
| 21:04.30 | Solver | you were arguing at cross-purposes - how's that? :) |
| 21:04.53 | nephfl | i dont know about "new" because one of the modules had pins bent like it was a pull, but i doubt they would send me 2 cards with a defective port 2, but it also works fine as an extension |
| 21:05.07 | KyleK | well he started it :) |
| 21:05.08 | nephfl | i can pick it up and call another extension and another extension can call it |
| 21:05.10 | Solver | haha |
| 21:05.23 | MaliutaLap | sets Solvers shell to /usr/bin/fortune |
| 21:05.28 | javb | Dahdi 2.1.0.4 would say "You d not appear to have the sources.... kernel installed" but 1.4 no problem |
| 21:05.34 | Solver | KyleK: I've known MaliutaLap for 15 years IRL and I am prepared to believe he started it ;) |
| 21:05.43 | MaliutaLap | rofl |
| 21:06.03 | Katty | so hungry. |
| 21:06.17 | Katty | pretzles didn't help |
| 21:06.27 | Katty | oh nice. less than 1g of fiber. |
| 21:06.29 | Katty | no wonder. |
| 21:06.42 | KyleK | I was going to guess over 35 |
| 21:08.04 | nephfl | this is great, it wasnt the card or the modules or software, it is a bad analog phone..oh my god... |
| 21:08.45 | [TK]D-Fender | hands nephfl his ClueBat (tm) |
| 21:08.50 | [TK]D-Fender | nephfl: You know what to do... |
| 21:09.15 | nephfl | not really, i wonder if it is the phone or if the old fashioned phone with a bell is requiring too much power |
| 21:09.44 | KyleK | mechanical bell? |
| 21:09.47 | nephfl | yeah |
| 21:09.48 | Katty | eppigy: i just had another snack :< |
| 21:09.56 | Katty | eppigy: that's my third snack so far this afternoon |
| 21:10.06 | nephfl | old fashioned red phone, whitehouse to kremlin style |
| 21:10.16 | KyleK | oooo |
| 21:10.19 | KyleK | rotary dial? |
| 21:10.32 | nephfl | would be, but there is no rotary or keys, it is a hotline |
| 21:10.51 | KyleK | yea dont plug that into VoIP stuff |
| 21:10.52 | [TK]D-Fender | nephfl: REN Killer! |
| 21:10.59 | nephfl | ? |
| 21:11.22 | KyleK | ~ren |
| 21:11.23 | infobot | hmm... ren is Ringer Equivalence Number - a telephone line can normally supply upto 4 REN, where a standard telephone/answering machine etc would equal 1 REN |
| 21:11.40 | nephfl | maybe i can disable the bell and put a ringer on it or something |
| 21:11.44 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 21:12.00 | nephfl | it shouldnt really ring anyway, it is the point that will be calling out |
| 21:12.13 | [TK]D-Fender | nephfl: Maybe you could replace it with a 10$ POS from your drug store that will work :) |
| 21:12.19 | nephfl | i wonder if i can pick up a ringer that will work at the local walmart |
| 21:12.28 | Katty | :< |
| 21:12.35 | KyleK | nephfl: thrift store? |
| 21:12.46 | KyleK | I bought a phone for $2 once |
| 21:12.49 | nephfl | i want to use the phone, because it looks cool, but i want it to ring too... |
| 21:13.39 | KyleK | nephfl: try and get an old phone for $2, and cut out almost everything cept the ringer on it and hide that inside the rotary? |
| 21:15.26 | *** join/#asterisk ELM2 (n=wow1602@mail.gotvoice.com) |
| 21:16.45 | Tene | I'm having a problem with musiconhold. 'moh show files' does list the files, and 'moh show classes' does show the 'default' class. |
| 21:17.15 | Tene | When I start a call to an extension that just runs MusicOnHold(), it says: started music on hold, class 'default', then says 'Stopped music on hold' |
| 21:17.29 | Tene | I've got debug and verbose turned way up |
| 21:17.47 | Tene | How can I get it to give me more information to find out what the problem is? |
| 21:20.04 | [TK]D-Fender | tene show us your files |
| 21:20.38 | *** join/#asterisk trentcreek (n=kvirc@200.94.227.117) |
| 21:20.59 | Tene | [TK]D-Fender: just musiconhold.conf? |
| 21:21.25 | rhassing | Tene, can you do a dahdi show status |
| 21:21.29 | [TK]D-Fender | Tene: That, the call attempt, and that actual files you intend to use. PASTEBIN the whoe mess |
| 21:21.31 | [TK]D-Fender | ~pb |
| 21:21.32 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 21:21.33 | [TK]D-Fender | ^^^^^^^^^ |
| 21:21.54 | rhassing | Tene, you might be missing your clocking |
| 21:22.16 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 21:22.16 | Tene | rhassing: ildcard TDM400P REV I Board 5 OK 0 0 0 CAS Unk YEL 0 db (CSU)/0-133 feet (DSX-1) |
| 21:23.04 | rhassing | Tene, Ok... I was just thinking :-) |
| 21:23.06 | *** join/#asterisk hi365 (n=hi365@94.159.178.51) |
| 21:23.48 | Tene | [TK]D-Fender: http://gist.github.com/115095 |
| 21:24.15 | Tene | [TK]D-Fender: ehich other configs? |
| 21:24.37 | [TK]D-Fender | Tene: Show me the ACTUAL files (ls dump |
| 21:25.00 | xpot-mobile | Question: any know what would cause a "407 Proxy Authentication Required" when calling another local extension? here is my output --> http://pastebin.com/d1c38a727 |
| 21:25.31 | *** join/#asterisk ddickenson_ (n=ddickens@rrcs-97-77-245-251.sw.biz.rr.com) |
| 21:26.09 | [TK]D-Fender | xpot-mobile: * wants it authed. Nothing irregular there |
| 21:26.57 | Tene | [TK]D-Fender: refresh the page. |
| 21:27.17 | xpot-mobile | [TK]D-Fender: if "insecure=very" is set why would it want it authed? I am already passing register info and it registers just fine. I only get the error when I try to call |
| 21:27.58 | [TK]D-Fender | Tene: Doesn't work that way |
| 21:28.10 | *** join/#asterisk cesar_CR (n=cesar@201.195.239.11) |
| 21:28.17 | trentcreek | What in asterisk is Producing these headers? http://pastebin.ca/1428153 |
| 21:28.51 | Tene | [TK]D-Fender: Can you explain what you're referring to? |
| 21:29.05 | javb | when installing dadhi tools, i het on "make" the error: No rule to make target "makeopts"... any idea?? |
| 21:29.22 | [TK]D-Fender | Tene: You CAN'T refresh. that is a fixed post an it gives you a NEW link |
| 21:29.33 | Tene | [TK]D-Fender: http://gist.github.com/115095 |
| 21:32.02 | [TK]D-Fender | Tene: I asked for FILE DUMP of your MOH FILES. |
| 21:32.15 | [TK]D-Fender | Tene: not CONFIGS. I want to see the SOUNDS FILES |
| 21:35.51 | Tene | [TK]D-Fender: uploading to http://pleasedieinafire.net/~tene/default/ slowly |
| 21:36.58 | [TK]D-Fender | Tene: Ok, we seem to have a basic comprehension problem here. I said "ls dump". Go to blooody *NIX CLI and "ls -la" the damn folder. |
| 21:37.13 | [TK]D-Fender | Tene: I didn't need to inspect every byte of the files |
| 21:37.38 | *** join/#asterisk DarkLogik (n=darklogi@76.73.51.195) |
| 21:37.49 | [TK]D-Fender | trentcreek: .ca is broken. Repaste |
| 21:38.26 | Tene | Yes, that was a misunderstanding. i've never heard someone refer to 'ls' as a dump. Here you go: http://nopaste.snit.ch/16624 |
| 21:38.35 | [TK]D-Fender | Tene: Next I don't recall * supporting .ogg <- |
| 21:39.02 | Tene | Huh, Okay. I must have misremembered. |
| 21:39.06 | [TK]D-Fender | Tene: I highy recommend you convert them to an * native format |
| 21:39.18 | Tene | OK |
| 21:39.52 | trentcreek | http://pastebin.com/d412293ab Another Link |
| 21:40.03 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
| 21:40.45 | [TK]D-Fender | trentcreek: So what are we supposed to be digging blindly for in there? |
| 21:41.19 | trentcreek | My question was, what in asterisk produces those headers |
| 21:41.22 | Tene | Ah, I turned the debug up more and I see format_vorbis complaining "Only monophonic OGG/Vorbis files are currently supported!" |
| 21:41.42 | KyleK | trentcreek: chan_sip? |
| 21:41.48 | *** join/#asterisk M1s3ry (n=M1s3ry@boromir.api-digital.com) |
| 21:41.51 | carrar | bits! |
| 21:41.53 | carrar | and bytes |
| 21:41.56 | carrar | of code! |
| 21:42.01 | [TK]D-Fender | trentcreek: Sure looks like a DIAL to me. |
| 21:42.01 | trentcreek | KyleK: thanks |
| 21:42.15 | trentcreek | yes it is...so I want to know where to look |
| 21:42.28 | [TK]D-Fender | And duh of course the SIP **channel driver** generates those |
| 21:42.31 | KyleK | Tene: can you knock off a channel without having to transcode? |
| 21:42.46 | [TK]D-Fender | trentcreek: Do you realize how ridiculously vague your quesiton is? |
| 21:43.13 | [TK]D-Fender | trentcreek: I can't tell if you think something is WRONG there, or want to know where in the source, or why you see it in your output or ANYTHING |
| 21:43.20 | trentcreek | Do you realise how it is not...whatt else would produce those headers? majic? |
| 21:43.24 | [TK]D-Fender | trentcreek: So stop the damn fishing expedition questions! |
| 21:43.27 | KyleK | hehehehehehehehehe |
| 21:43.44 | KyleK | trentcreek, [TK]D-Fender: I love magicjack |
| 21:43.46 | KyleK | runs off |
| 21:43.49 | [TK]D-Fender | trentcreek: How do we know you aren't referring to 2 specific lines in that output? |
| 21:43.57 | [TK]D-Fender | trentcreek: FFS be specific |
| 21:44.07 | *** join/#asterisk telecos (n=sergio@134.167.219.87.dynamic.jazztel.es) |
| 21:44.11 | trentcreek | i asked for the HEADERS , not any part of any line. |
| 21:44.25 | [TK]D-Fender | trentcreek: thats 18 lines of HEADERS! |
| 21:44.43 | [TK]D-Fender | trentcreek: More vague crap. WHAT ABOUT THEM? |
| 21:44.53 | trentcreek | exactly...so WHAT PRODUCED those headers. and KyleK answered |
| 21:45.02 | [TK]D-Fender | trentcreek: Chap_sip generates the headers. The end. |
| 21:45.02 | carrar | Asterisk! |
| 21:45.05 | KyleK | i uh, answered the question Fender |
| 21:45.10 | carrar | tk, be vauge back :) |
| 21:45.23 | carrar | A SIP Device |
| 21:45.31 | trentcreek | thanks KyleK |
| 21:46.18 | KyleK | yw |
| 21:48.16 | [TK]D-Fender | :/ |
| 21:49.06 | timeshell_atwork | http://mikecarano.com/startrek.html |
| 21:51.21 | haryv | Hi TK. I dont know if you or somone else could give me a idea why my asterisk box does not respond when the firewall is off. We do this over night because the extra power usage is not nessesary. I can display some log files at pastebin. |
| 21:52.39 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 21:53.07 | Nilzao | see ya |
| 21:53.09 | *** part/#asterisk Nilzao (n=nils@200-168-146-103.dsl.telesp.net.br) |
| 21:53.16 | timeshell_atwork | Well, that |
| 21:53.22 | timeshell_atwork | ... is an interesting question. |
| 21:53.27 | haryv | Tk, do you know if 1.6 is considerably more stable then 1.4? |
| 21:53.32 | haryv | What is? |
| 21:53.49 | timeshell_atwork | Is your firewall on the same server as your asterisk? |
| 21:53.57 | haryv | no...not yet. |
| 21:54.10 | timeshell_atwork | Then you have a networking configuration problem. |
| 21:54.20 | timeshell_atwork | Should have nothing to do with asterisk. |
| 21:54.53 | haryv | Network configuration possibly. I can still log into asterisk but then, it will run again. |
| 21:55.25 | haryv | DNS is set corretly. I can probebly refer this to the forums. |
| 21:55.50 | timeshell_atwork | Dude, your question is incredibly vague and sounds like it really has nothing to do with asterisk. |
| 21:56.15 | *** join/#asterisk b14ck (n=comradeb@72.37.252.50) |
| 21:56.19 | timeshell_atwork | I don't want to sound unhelpful, but firewall is on it works, firewall is off it doesn't. Well, 2 and 2 here. |
| 21:56.39 | Qwell | if the firewall is off...what is passing packets? |
| 21:57.41 | haryv | I have the firewall on the asterisk box off. I noticed with it on, it prevents the phones from registering. |
| 21:58.01 | Qwell | ...and? |
| 21:58.16 | timeshell_atwork | Well, it would if you don't open the ports asterisk requires. |
| 21:58.24 | haryv | I know |
| 21:58.25 | haryv | :) |
| 21:58.36 | Qwell | with it "off" (your default policy is probably REJECT) it doesn't respond at all... |
| 21:58.41 | Qwell | so fix your firewall. #iptables |
| 21:59.15 | Nugget | pf > iptables :) |
| 21:59.29 | drmessano | Trolling > Idling |
| 21:59.30 | Qwell | Nugget: openssh exploit. |
| 21:59.37 | Qwell | You lose. :P |
| 21:59.48 | timeshell_atwork | I think Qwell's question was referring to the implication of what you said referring to an separate firewall server. |
| 21:59.52 | Qwell | (netbsd exploit trumps all trolling) |
| 21:59.57 | Nugget | heh |
| 21:59.59 | drmessano | heh |
| 22:00.08 | Nugget | what do you suggest, switching to gnussh? oh, wait. :P |
| 22:00.14 | drmessano | So pwning is better than pwning? |
| 22:00.17 | drmessano | That pwns |
| 22:00.18 | Qwell | heh |
| 22:00.24 | haryv | This is be our main FW and asterisk box. Just trying to cut down on the watts used on a 24/7 basis. |
| 22:00.27 | Qwell | Nugget: so what is the recent hole they found? |
| 22:00.53 | Qwell | oh, wow |
| 22:00.56 | timeshell_atwork | haryv Your confusing me. Your original statement said that asterisk didn't work when the FW was off |
| 22:01.12 | drmessano | haryv: You've wasted the $5 a month you would have spent on hot air being blown across IRC. Keep the FW box up and move along now |
| 22:01.17 | timeshell_atwork | Now, you want it on the asterisk box and it doesn't work there because the ports aren't open |
| 22:01.30 | timeshell_atwork | lol |
| 22:01.30 | Nugget | I'm not aware of a recent openssh exploit. There was an openssl issue a month or so ago |
| 22:01.38 | drmessano | Unless you're trentcreek and it costs you $100 a month to run a PC at home |
| 22:01.45 | Qwell | http://www.theregister.co.uk/2009/05/19/open_ssh_hack/ |
| 22:01.49 | drmessano | In whatever 3rd world country he lives in |
| 22:01.56 | Qwell | it from theregister.co.uk, so take it with a grain of salt |
| 22:01.57 | Nugget | looks |
| 22:01.58 | drmessano | Louisiana I think |
| 22:01.59 | haryv | time, I have a seperate fw/nat box. That box, we turn off late at night or power outage. Some times, asterisk decides to not run in cases likethat. |
| 22:02.16 | Qwell | haryv: If said box is *powered off*, how are packets passing through it? |
| 22:02.18 | drmessano | haryv: leave the box up and running or fix it |
| 22:02.30 | Nugget | ahh theregister. |
| 22:02.32 | drmessano | Qwell: THE NIC IS LIT |
| 22:02.45 | drmessano | Qwell: ITS FLASHY |
| 22:02.50 | Qwell | Nugget: their interpretation may be flawed, but it's a real issue |
| 22:02.51 | drmessano | Qwell: == Passing |
| 22:02.55 | timeshell_atwork | harvy, Draw out your network path between your phones, asterisk server, switches and fw/nat box. |
| 22:02.59 | timeshell_atwork | On a piece of paper. |
| 22:03.01 | Nugget | looks like they're reporting on the issue from last november |
| 22:03.03 | Nugget | http://www.cpni.gov.uk/Docs/Vulnerability_Advisory_SSH.txt |
| 22:03.08 | carrar | FLASHY!! |
| 22:03.09 | Qwell | Nugget: ahh |
| 22:03.10 | trentcreek | bah...Its freaking hot n humid where I live..electricity aint cheap, and on top of running a computer, itr causes the A/C to run longer. Its an old as central unit made when energy was still cheap |
| 22:03.14 | drmessano | Use multicolored crayons please.. I am black/white color blind |
| 22:03.19 | timeshell_atwork | haryv, Draw out everything down to the wire. |
| 22:03.20 | Qwell | yeah, looks like it |
| 22:03.29 | timeshell_atwork | Then cross out the item that disappears when you turn it off. |
| 22:03.34 | timeshell_atwork | I think you'll have your own answer. |
| 22:03.41 | drmessano | timeshell_atwork: NO |
| 22:03.54 | drmessano | timeshell_atwork: Tell him to put a Mr Yuck sticker over what disappears |
| 22:03.56 | timeshell_atwork | drmessano: NO? |
| 22:04.03 | carrar | Nuggest, hahah OpenSSH 4.7p1 |
| 22:04.03 | timeshell_atwork | lol!! |
| 22:04.04 | drmessano | timeshell_atwork: Far easier |
| 22:04.09 | haryv | Let me make my self clear. Asterisk box has firewall that is off. It is off because it will NEED to be configured to allow rtp/sip ect packets pass though its interfaces. The main fw is on. When that server is powered off, I get a no responce from asterisk. |
| 22:04.26 | drmessano | What is ect? Is that RTP for IAX3? |
| 22:04.29 | Qwell | Nugget: which is it that has the insane security record? net or open? |
| 22:04.33 | drmessano | is confuxored |
| 22:04.33 | [TK]D-Fender | [18:03]<drmessano>Use multicolored crayons please.. I am black/white color blind <- black & white aren't "colours" :0 |
| 22:04.38 | timeshell_atwork | haryv You need to follow your network path. |
| 22:04.42 | Nugget | open |
| 22:04.45 | Qwell | ahh |
| 22:05.06 | timeshell_atwork | haryv Obviously your fw server is somewhere between your asterisk server and your phones, either physically or logically. |
| 22:05.13 | drmessano | [TK]D-Fender: No, they are NOT.. especially when you cant effing SEE THEM |
| 22:05.19 | Qwell | did they stop advertising "Only 4 holes in the default config in 87 years!" ? |
| 22:05.29 | timeshell_atwork | haryv We cannot debug that for you. You need to find out where your network path is broken. |
| 22:05.31 | Nugget | freebsd focuses on managability and performance, openbsd focuses on security, and netbsd focuses on portability. loosely speaking, of course. |
| 22:05.34 | carrar | Nugget, 4.7 welcome to Sept 2007 |
| 22:05.36 | Qwell | or whatever absurd (if not impressive..) claims? |
| 22:05.39 | haryv | ugg |
| 22:05.40 | Nugget | "Only two remote holes in the default install, in a heck of a long time! |
| 22:05.48 | drmessano | HAHAH |
| 22:05.59 | haryv | Im going to deal with the iptables issue. |
| 22:06.00 | [TK]D-Fender | Qwell: "Exaggerating statisitics for 53.4 years!" |
| 22:06.04 | Qwell | oh, right... openbsd is also Theo |
| 22:06.09 | drmessano | I remember when it was "ONE HOLE IN 37 YEARS"... now its "A few holes in a heck of a while!!" |
| 22:06.10 | Qwell | [TK]D-Fender: in this case, it's actually accurate |
| 22:06.16 | Qwell | drmessano: yeah, heh |
| 22:06.16 | drmessano | lame |
| 22:06.28 | Nugget | eyes carrar confused |
| 22:06.33 | *** join/#asterisk outtolunc (n=me@c-71-198-197-201.hsd1.ca.comcast.net) |
| 22:06.37 | timeshell_atwork | haryv Also, it appears that even if you turn on the fw on your asterisk box, turning off the fw server will still cause a problem as it apparently is part of your network route. |
| 22:06.57 | carrar | Nugget you paste is OLD version of SSH, time to upgrade if you are using something 2 years old |
| 22:06.58 | drmessano | OpenBSD has cute mascots and swag.. I've no clue how to use it, however |
| 22:07.11 | Nugget | what paste? |
| 22:07.18 | timeshell_atwork | haryv SO, if you don't want it to be a part of your network route, you likely need to redesign your network. |
| 22:07.20 | carrar | <Nugget> http://www.cpni.gov.uk/Docs/Vulnerability_Advisory_SSH.txt |
| 22:07.20 | Qwell | carrar: it was a flaw of the protocol, if I'm not mistaken.. |
| 22:08.14 | drmessano | draws a line between the router and the air conditioner |
| 22:08.14 | carrar | still thats a old version |
| 22:08.14 | drmessano | Like that? |
| 22:08.14 | Nugget | carrar: the exploit is valid through 5.1 |
| 22:08.14 | Qwell | carrar: that's just what it was verified against |
| 22:08.14 | Nugget | please try to keep up |
| 22:08.14 | carrar | 5.2? |
| 22:08.15 | carrar | is current |
| 22:08.15 | Nugget | no, 5.2 fixes it |
| 22:08.15 | timeshell_atwork | drmessano : NO, put a MrYuck sticker on the Air Conditioner FIRST |
| 22:08.15 | hff135 | does anyone know if zaptel timing issues can affect call quality where the call is not transcoded and there is no PSTN interface on the asterisk box? |
| 22:08.16 | drmessano | thinks carrar just ate a butt nugget |
| 22:08.16 | carrar | who isn't keeping current with openssh? |
| 22:08.26 | Nugget | you're not keeping up with the conversation |
| 22:08.33 | Nugget | consult your scrollback |
| 22:08.37 | carrar | heh |
| 22:08.39 | haryv | timeshell, the only case where there would be a disruption is my DIDs. But not my phones on my local lan network. The phones work find for hours with fw off. Its when I leave shorter or longer periods, then I would use my phone and the asterisk box is not responding. Some times, I log into asterisk, and it will now allow the phones to respond. |
| 22:08.45 | Qwell | hff135: what kind of quality issues? |
| 22:08.57 | drmessano | timeshell_atwork: No, I need the Air conditioner OFF, the router ON, the asterisk box OFF, the firewall ON STANDBY, and I need to be able to make calls |
| 22:09.04 | drmessano | timeshell_atwork: Now help me, PLEASE |
| 22:09.05 | carrar | I hate negotiating with my scroll back |
| 22:09.06 | haryv | Its at times a hit and miss issue. |
| 22:09.16 | hff135 | the quality issues are dropped words |
| 22:09.26 | drmessano | Wait, I need the air conditoner off |
| 22:09.30 | drmessano | No, on |
| 22:09.32 | timeshell_atwork | haryv Then it sounds like a networking issue on the asterisk server itself. |
| 22:09.33 | haryv | dr hehe |
| 22:09.35 | drmessano | shit.. the router is off |
| 22:09.41 | [TK]D-Fender | carrar: I do not negotiate with scrollback! |
| 22:09.43 | Qwell | hff135: loosely quoting Alex Balashov - do you live in soviet russia? |
| 22:09.47 | carrar | heh |
| 22:09.49 | haryv | drmessano, dont be a doof :) |
| 22:09.55 | Qwell | (his posts are so epic..) |
| 22:10.00 | carrar | Can't I just pick random things out and comment on theM? :) |
| 22:10.07 | timeshell_atwork | haryv I've seen some linux OS's not like it if it can't talk to its default gateway and it gets hung up. |
| 22:10.18 | hff135 | i don't live in soviet russia. i live in canada ... so the climate is about the same |
| 22:10.20 | drmessano | haryv: Doof? You're the one who wants to turn, of all things, your FIREWALL off at night and STILL pass DATA |
| 22:10.21 | timeshell_atwork | I'd bet you're using RH, CentOS or Fedora. |
| 22:10.26 | drmessano | haryv: THUD THUD man |
| 22:10.38 | Qwell | hff135: explain a little better, if you can. is it always happening throughout the call? does it sound jittery, or are they just flat out dropped? |
| 22:10.46 | haryv | dr, we are on a local network with pots. We dont need dids late at night. |
| 22:11.03 | drmessano | haryv: That statement made no sense |
| 22:11.17 | drmessano | haryv: You have POTS, but dont need DIDs at night? WTF does that even mean? |
| 22:11.22 | hff135 | the quality is generally fine. however, the calls go bad from time to time. once in a while, words are dropped. both sides can report this problem |
| 22:11.25 | Qwell | drmessano: he's a chef |
| 22:11.26 | hff135 | it's not really jittery |
| 22:11.38 | [TK]D-Fender | Qwell: But one of these days he'll MAKE IT BIG!! |
| 22:11.39 | drmessano | Qwell: Hes David Copperfield |
| 22:11.45 | hff135 | it doesn't happen on every call. however it happens throughout the day, every day |
| 22:11.46 | Qwell | hff135: how many concurrent calls? over what kind of connectivity? |
| 22:12.02 | hff135 | we're just testing right now. 1 concurrent call |
| 22:12.07 | Qwell | SIP phones, to Asterisk, through an ITSP? |
| 22:12.12 | drmessano | Qwell: I need to turn off the VO and leave the IP on at night |
| 22:12.16 | haryv | Dr, I dont need incomming calls late at night. 95 percent of our calls are though the cards. |
| 22:12.42 | timeshell_atwork | haryv It sounds more like you have an issue with the operating system on your asterisk server. |
| 22:12.43 | drmessano | ..... |
| 22:12.49 | timeshell_atwork | Look at the networking there |
| 22:13.10 | drmessano | Sounds to me like a TTY >< Chair disconnect issue |
| 22:13.10 | hff135 | the asterisk box is located in our data center. internet is good. having said that, i'm looking into all possible issues so we are going to look into network as the cause also |
| 22:13.19 | timeshell_atwork | gotta go |
| 22:13.21 | timeshell_atwork | later |
| 22:13.24 | Qwell | hff135: that's where I would start looking, yes |
| 22:13.25 | hff135 | when i run zttest, i get 99.95%. this is lower than it should be |
| 22:13.30 | haryv | Did not mention, there is a bug crash. So need to look at that also. |
| 22:13.38 | hff135 | i'm wondering if zaptel timing issues are causing this |
| 22:13.40 | Qwell | hff135: timing isn't really important, and it wouldn't cause loss of packets |
| 22:15.12 | hff135 | so shoudl i be worried about that zttest score? |
| 22:15.19 | Qwell | no |
| 22:15.27 | hff135 | ok. that's helpful |
| 22:20.08 | nephfl | how can i get the call file to connect to a MeetMe ? it doesnt like it as an app or in a context |
| 22:20.36 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 22:27.25 | *** join/#asterisk paulius (n=paulius@unaffiliated/paulius) |
| 22:27.40 | paulius | I need echo cancellation tips and tricks for the SPA-3102, [ep[;e/ |
| 22:28.02 | paulius | And yes I've read the articles that some people have gave me. Are there any more tips? |
| 22:28.41 | nephfl | anybody know how to connect to meetme from a call file? |
| 22:28.55 | jameswf | nephfl: practice |
| 22:29.04 | nephfl | practice? |
| 22:29.57 | jameswf | ~echo |
| 22:29.58 | infobot | well, echo is an issue which can be best fixed using this link: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-8-SECT-5.html, or fixed with fxotune: http://www.voip-info.org/wiki/view/Asterisk+fxotune, or best fixed by troubleshooting your pci bus: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting |
| 22:31.36 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 22:33.31 | Qwell | jameswf: I know. |
| 22:33.39 | KyleK | hmm |
| 22:33.48 | Qwell | jameswf: how do I know? because there's nothing better, and it would be dumb not to :P |
| 22:33.55 | KyleK | paulius: I haven't had any issues with echo cancellation with my spa3102 |
| 22:34.10 | paulius | KyleK: Well it's not echo cancellation, it's simply echo. |
| 22:34.26 | paulius | And people recommended me a bunch of articles, and I've followed their advice. |
| 22:34.38 | paulius | Tweaking the settings improved the situation a bit |
| 22:34.46 | paulius | but it's far from being a usable system |
| 22:34.49 | KyleK | ah, i haven't read anything on echo :) |
| 22:35.26 | paulius | so you're running your thing with default settings and not getting any echo at all? |
| 22:36.15 | KyleK | I've gotten a bit of echo calling my sister but i figure thats the cell system being inconsistant |
| 22:37.19 | drmessano | KyleK: A slight tweak in gain would likely fix that.. I bet she either speaks soft which causes you to talk loud, or she speaks loud |
| 22:37.41 | drmessano | and it exaggerates a slight mismatch |
| 22:38.00 | KyleK | yea shes quiet, especially compared to her friends |
| 22:38.34 | drmessano | Yeah.. I have had that too... I speak loud, so I had to make a tweak even after I had it set "correctly" |
| 22:39.22 | [TK]D-Fender | nephfl: Question doesn't make a lot of sense... |
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| 22:48.20 | nephfl | what do you mean it doesnt make sense? |
| 22:48.34 | nephfl | i need a call file to ring a channel and connect it to a meetme when it picks up |
| 22:48.57 | nephfl | in the call file i have tried MeeMe(1) as an application that doesnt work |
| 22:49.11 | [TK]D-Fender | nephfl: When it picks up go dump it into an EXTEN that puts it in a mettme |
| 22:49.13 | trentcreek | nephfl: maybe you are referring to an AGI script |
| 22:49.20 | nephfl | i have tried a context where S sends it to MeetMe |
| 22:49.42 | nephfl | no, i was going to use an agi script to move a couple of call files |
| 22:49.43 | [TK]D-Fender | nephfl: If you've tried stuff and failed, show us what you did so we can show where it went wrong |
| 22:49.54 | nephfl | ok |
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| 23:05.49 | *** join/#asterisk hesco (n=hesco@24.99.160.121) |
| 23:12.42 | jameswf | infobot: drop table; |
| 23:12.48 | jameswf | bahh |
| 23:13.05 | [TK]D-Fender | ~die |
| 23:13.06 | infobot | ACTION takes two shots to the head and crumples to the ground, lifeless. |
| 23:13.16 | [TK]D-Fender | ~die |
| 23:13.17 | infobot | ACTION takes two shots to the head and crumples to the ground, lifeless. |
| 23:13.21 | [TK]D-Fender | hrm |
| 23:13.28 | [TK]D-Fender | ~end |
| 23:13.28 | infobot | well, end is near. |
| 23:14.10 | [TK]D-Fender | ~kill |
| 23:14.22 | [TK]D-Fender | Yeah, I forgot the really funny one |
| 23:14.34 | [TK]D-Fender | ~killall |
| 23:14.34 | infobot | killall is, like, a bad idea on non GNU platforms, or ok on BSD, too |
| 23:16.25 | jameswf | ~~ |
| 23:16.26 | infobot | ~ is probably the key |
| 23:16.29 | jameswf | ~~~ |
| 23:16.48 | jameswf | someone took infobot's humor away |
| 23:17.39 | paulius | ~cisco |
| 23:17.40 | infobot | cisco makes the routers that move the pr0n across the internet at a very high rate of exchange. a company full of patent protected punks!, or <reply>Cisco phones are expensive crap which should be avoided with extreme prejudice |
| 23:17.53 | paulius | ~richard stallman |
| 23:17.53 | infobot | methinks richard stallman is known as RMS |
| 23:18.16 | paulius | I think that infobot thinks that richard stallman is the world's savior |
| 23:20.07 | KyleK | ~satan |
| 23:20.07 | infobot | methinks satan is in a7r's pants |
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| 23:23.27 | *** part/#asterisk nullable_type (n=nullable@hq.verbx.net) |
| 23:26.06 | MaliutaLap | kicks dahdi in the balls |
| 23:26.18 | MaliutaLap | can't see why it started working all of a sudden |
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| 23:31.58 | jameswf | ÙÙ ÙدÙ٠اÙرؤÙس اÙÙÙÙÙØ© |
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