| 00:02.28 | *** join/#asterisk DelphiWorld (n=Miranda@41.201.100.74) |
| 00:02.36 | DelphiWorld | hello all |
| 00:02.50 | DelphiWorld | how to compare libiax2 with iaxcli? |
| 00:02.50 | matt_d | hi delphi |
| 00:02.59 | DelphiWorld | matt_d: hi |
| 00:03.27 | DelphiWorld | i'm building a SIP/IAX softphone |
| 00:03.37 | DelphiWorld | and i need a recommanded IAX2 implementation |
| 00:03.45 | *** join/#asterisk fnord0 (n=fnord0@unaffiliated/fnord0) |
| 00:04.17 | matt_d | why use iax? |
| 00:04.18 | box2 | DelphiWorld: try looking at iaxcomm |
| 00:05.00 | DelphiWorld | box2: iaxcom is allready developed a a softphone, i need a library |
| 00:05.03 | box2 | because iax doesn't require 90,000 open ports |
| 00:05.10 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 00:05.11 | DelphiWorld | matt_d: i must use IAX, ISP blocking SIP |
| 00:05.21 | matt_d | DelphiWorld: oh.. just curious. |
| 00:05.25 | matt_d | box2: are you DelphiWorld? |
| 00:05.33 | box2 | i am not |
| 00:05.35 | DelphiWorld | blam algeria telecom |
| 00:05.42 | matt_d | cool :) |
| 00:06.06 | matt_d | box2: that is one thing annoying about SIP |
| 00:06.14 | box2 | yes, yes it is |
| 00:06.30 | box2 | DelphiWorld: iaxcomm is open source, you can look at what libraries they use to interface with IAX |
| 00:06.34 | DelphiWorld | just question: |
| 00:06.39 | manxpower | SIP requires 2 ports per call. |
| 00:06.49 | DelphiWorld | iax2 is in continou developmant or stoped? |
| 00:07.20 | manxpower | DelphiWorld: no significant protocol level changes in a long time |
| 00:07.36 | DelphiWorld | manxpower: ho |
| 00:07.55 | DelphiWorld | request from digium to continu IAX2 to IAX2.1 or IAX3 |
| 00:08.29 | matt_d | request's that ChanSpy's whisper feature is fixed :) |
| 00:08.30 | matt_d | hehe |
| 00:08.38 | DelphiWorld | matt_d: :D |
| 00:08.43 | DelphiWorld | i think i use libiaxclient |
| 00:08.58 | DelphiWorld | for SIP Stack, i use OpenSipStack |
| 00:12.08 | DelphiWorld | i can't found libiaxclient could anyone give me the URL? |
| 00:12.46 | matt_d | for what platform? |
| 00:13.16 | *** join/#asterisk ingenius (n=alektro@186.136.25.189) |
| 00:13.22 | *** join/#asterisk coppice (n=chatzill@52.204.17.210.dyn.pacific.net.hk) |
| 00:13.23 | DelphiWorld | matt_d: unfortunatly Win32 :D |
| 00:13.28 | matt_d | which platform |
| 00:13.39 | matt_d | cygen, or wahtever its called? |
| 00:13.39 | DelphiWorld | coppice: hi the T.38 master |
| 00:13.53 | DelphiWorld | matt_d: i don't know that, i use window |
| 00:13.56 | DelphiWorld | matt_d: i don't know that, i use windows |
| 00:14.06 | DelphiWorld | is called libiaxclient |
| 00:14.27 | *** join/#asterisk ingenius (n=alektro@186.136.25.189) |
| 00:14.27 | matt_d | yes, but don't you have to run asterisk under cygwin ? the linux enviolrment for windows? |
| 00:14.45 | DelphiWorld | matt_d: ok understand! |
| 00:14.47 | box2 | i know X-Lite supports IAX on windows |
| 00:14.50 | DelphiWorld | matt_d: no i use ubuntu for asterisk |
| 00:15.02 | box2 | i don't know what libraries it uses for IAX client though |
| 00:15.03 | DelphiWorld | matt_d: asterisk don't work wel in win32 except for 1.2 |
| 00:15.14 | matt_d | anyway, iaxclient.sourceforge.net is the website for the lib u want |
| 00:15.17 | DelphiWorld | box2: libiaxclient |
| 00:15.19 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:15.27 | DelphiWorld | matt_d: thanks |
| 00:17.38 | DelphiWorld | is checking out iaxclient from svn |
| 00:18.08 | matt_d | np |
| 00:18.37 | DelphiWorld | box2: IaxCom use iaxclient and is included in the SVN repository, thank you for recommanding it |
| 00:19.00 | box2 | np |
| 00:19.15 | DelphiWorld | box2: got a .Net wrapper! |
| 00:19.21 | DelphiWorld | this will help me :D |
| 00:19.30 | matt_d | please don't say .Net |
| 00:19.31 | box2 | lol cool |
| 00:19.36 | matt_d | hehe j/k :) |
| 00:19.51 | DelphiWorld | matt_d: why? |
| 00:20.08 | Dotnet_mono | :d |
| 00:20.22 | matt_d | becuase i don't like that framework :) |
| 00:20.24 | matt_d | just joking around |
| 00:20.43 | DelphiWorld | matt_d: what you like? |
| 00:20.51 | matt_d | Cocoa :) |
| 00:21.28 | DelphiWorld | request C/C++ courss from matt_d |
| 00:22.17 | matt_d | I like Obj-C with Cocoa.. In fact, u can use Cocoa with Ruby now |
| 00:22.46 | box2 | my best friend uses only Ruby now and it makes me cry myself to sleep at night :( |
| 00:22.57 | DelphiWorld | matt_d: unfortunatly i don't know C/C++ developmant only Delphi/VB :D |
| 00:23.23 | matt_d | box2: i started using ruby this year. i love it! in fact im starting a FastAGI server written all in Ruby. |
| 00:23.31 | box2 | :((((( |
| 00:23.43 | box2 | infidels! |
| 00:23.45 | matt_d | DelphiWorld: as in Delphi u mean Pascal (i dont know sh*t about window) . then going to C will be easy for you. |
| 00:23.49 | DelphiWorld | what about LUA? |
| 00:24.01 | box2 | Lua is pretty good |
| 00:24.02 | matt_d | DelphiWorld: well, the language of C. there are some differences as far as pointers, etc. |
| 00:24.30 | matt_d | When i first learned computer programming I started with BASIC, Pascal and C. |
| 00:24.33 | matt_d | C was the easiest. |
| 00:24.48 | DelphiWorld | LUA is very good, i'm writing IVR's using it with Freeswitch |
| 00:25.16 | DelphiWorld | lua is 100% embeddable |
| 00:25.32 | coppice | When i first learned computer programming there was no Pascal or C :-\ |
| 00:25.49 | box2 | Lua is awesome, i love it |
| 00:25.54 | box2 | very fast to make good stuff |
| 00:26.23 | matt_d | coppice: just fortran, smalltalk and cobol? |
| 00:26.25 | DelphiWorld | box2: after it i love JS |
| 00:26.58 | coppice | there was no smalltalk. that's newer than C |
| 00:27.12 | box2 | i like JS only because they wrote Quake Live with it heh |
| 00:27.15 | DelphiWorld | i think coppicelike only C |
| 00:27.22 | matt_d | coppice: oh thats right. st is newer than C.. then COBOL and fortran then ? :) |
| 00:27.26 | florz | _C_ and _easy_? |
| 00:27.34 | matt_d | coppice: i learned COBOL but never actually used it. |
| 00:27.58 | coppice | that's the best amount of use of cobol |
| 00:28.11 | matt_d | coppice: i agree |
| 00:28.20 | DelphiWorld | coppice: use only your T.38 stuf and ignor programing :D |
| 00:28.26 | coppice | the first language I learned was fortran ii |
| 00:30.17 | *** join/#asterisk propellerhead (n=yogurt2u@host126.190-30-35.telecom.net.ar) |
| 00:31.55 | matt_d | box2: the game? or the servers? |
| 00:32.03 | box2 | the game client |
| 00:32.13 | box2 | you play it directly from your browser |
| 00:32.19 | box2 | is> ELITE< |
| 00:32.32 | matt_d | box2: oh okay .. i used to love Java and JS.. but got frustrated with that language.. i don't know why .. hehe |
| 00:35.46 | manxpower | *heart* COBOL. |
| 00:36.01 | manxpower | RPG-II is what I hated. |
| 00:36.26 | *** join/#asterisk rossand (n=rossand@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 00:36.42 | manxpower | Modula-2 was the first language I really loved. |
| 00:37.06 | DelphiWorld | never heare about Modula-2 |
| 00:37.06 | matt_d | A friend of mine created a programming language (not an interperted language, but linkable lanauge) that was very powerful and simple. he made it to teach kids how to program. very impressive language. had a built in framework to have full control over graphics, io, etc.. was pretty cool. |
| 00:37.42 | DelphiWorld | matt_d: tel it to release it in Open Source License |
| 00:37.47 | manxpower | matt_d: Was it called Logo 2? |
| 00:37.52 | DelphiWorld | good night all |
| 00:37.53 | coppice | logo used to be popular for teaching small kids |
| 00:38.09 | *** join/#asterisk KyleK (n=Kyle@allspark.shadowmage.org) |
| 00:38.09 | matt_d | manxpower: no.. i forgot waht he called it.. Logic something or other. |
| 00:38.11 | manxpower | coppice: *nod* It was the 2nd language I learned. |
| 00:38.20 | DelphiWorld | coppice: tomoro i will discuce with you about a Virtual printer driver for T.38 Faxing |
| 00:38.42 | *** part/#asterisk DelphiWorld (n=Miranda@41.201.100.74) |
| 00:39.06 | matt_d | wasn't RPG-II similiar to asm? |
| 00:39.47 | drmessano | I created my own religion.. Was based on people throwing themselves in volcanoes in worship of the great drmessano. Our followers were so devoted that we basically ran out of money due to lack of contributions. I had already begun rewriting our scripture, known as "The Bagel", but by the time "Bagel II: Electric Boogaloo" went to the printers, it was all over. |
| 00:40.12 | manxpower | matt_d: It is a reporting language. It was mainly based on which column the "command" was in determined what it did. It was so long ago all I remember is those horrid little forms you had to write the code on. |
| 00:40.14 | matt_d | drmessano: darn, i missed out on joining then. eh? |
| 00:40.30 | coppice | RPG-II was a report generator |
| 00:41.01 | manxpower | Rinky-dink Paper Generator is what we called it. |
| 00:41.09 | matt_d | i have to say that i don't have "fun" programming until I met Ruby. |
| 00:41.12 | matt_d | now its fun again. |
| 00:41.20 | drmessano | matt_d: Feel free to worship.. But please, let me send you a PDF of Bagel II. Chapter 1 is a total rewrite where "Throw yourself in a volcano" is replaced with "Give drmessano a bunch of money and throw yourself in a volcano". |
| 00:41.25 | drmessano | Its the small things.. |
| 00:41.44 | matt_d | well.. Obj-C 2 is fun too! |
| 00:41.56 | matt_d | drmessano: see that will fix it.. money before you throw yourself in the volcano :) |
| 00:42.16 | drmessano | matt_d: Oversights are a bitch |
| 00:42.27 | coppice | Obj-C is a lot nicer to use than C++ |
| 00:42.57 | matt_d | coppice: oh yeah! much better object model. makes more sense to me. |
| 00:43.29 | coppice | though I haven't used Obj-C for 16 or 17 years, since I last used a Next |
| 00:45.05 | box2 | NeXTSTEP!@#*)@#$ |
| 00:45.37 | matt_d | u will like Obj-C 2.0 then . better garbage collection, better performance ... |
| 00:45.38 | drmessano | manxpower: I would love to finish porting Asterisk to Logo. I need to find some way, when a call is put in a queue, to get past the turtle goin all screwy-like and producing the exact pattern needed to induce human vomiting. I probably need to look at better use of the screen blanker. |
| 00:45.56 | KyleK | damn i was hoping skype for asterisk wouldn't be priced for businesses |
| 00:46.53 | matt_d | brb wife is home with FOOD! :) |
| 00:48.46 | drmessano | KyleK: Could have been worse.. $66.60 |
| 00:48.54 | drmessano | MARK OF THE SKYPE |
| 00:49.02 | leifmadsen | heh |
| 00:49.58 | raden_work | how can i make my linksys wireless router just part of my network so i can use my VOIP cordless phones it plugs into our netgear router |
| 00:50.07 | raden_work | i want the netgear to assign all ips |
| 00:50.08 | drmessano | I'm sure digium has some badass dev costs to make up for.. I don't think they're pricing it to get rich |
| 00:51.54 | drmessano | Too bad they dont offer "first channel free" like the fax app :) |
| 00:52.50 | KyleK | haha first channel cheap would also make me very happy :) |
| 00:53.27 | drmessano | They DID price it cheaper than analog card, per port |
| 00:53.28 | coppice | how much is it? |
| 00:53.34 | drmessano | $66, per channel |
| 00:53.56 | coppice | is that discounted from $66.6? |
| 00:54.06 | drmessano | Youre late |
| 00:54.14 | drmessano | [20:49] <drmessano> KyleK: Could have been worse.. $66.60 |
| 00:54.14 | drmessano | [20:49] <drmessano> MARK OF THE SKYPE |
| 00:54.52 | drmessano | You get points for thinking the same thing, though |
| 00:57.32 | drmessano | I guess any card/channel driver is worth what it gives you access to.. I havent looked at the value of Skype's termination and origination.. Maybe its worth it |
| 00:58.43 | coppice | there are various free options, but they are clunky. therefore it seems the value of clunkiness in today's market is $66 |
| 00:59.22 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 00:59.22 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-snefhlaxpaeqpxrx) |
| 00:59.52 | drmessano | coppice: LOL |
| 01:00.05 | drmessano | Thats awesome |
| 01:01.28 | drmessano | $3 a month unlimited US and Canada outbound |
| 01:01.45 | drmessano | $5.95 to pick a country outside that |
| 01:01.54 | drmessano | $12.95 unlimited world |
| 01:02.45 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 01:04.16 | drmessano | If you purchase a monthly subscription, its $30 a year for their unlimited inbound+DID |
| 01:04.45 | leifmadsen | that's not too shabby |
| 01:05.01 | drmessano | Damn under $6 a month |
| 01:05.11 | drmessano | If you get US and Canada only + DID |
| 01:06.07 | drmessano | $11 a month the first year if you add the cost of the Skype license :) |
| 01:06.33 | leifmadsen | not too shabby for phone |
| 01:07.41 | drmessano | Go to Walmart and buy a Skype phone, $30+ |
| 01:07.51 | drmessano | So youre halfway to the license for * |
| 01:08.01 | drmessano | and it needs a tether to your PC |
| 01:08.05 | leifmadsen | yep |
| 01:08.08 | drmessano | and the ugliness of the skype client, etc |
| 01:08.11 | leifmadsen | and no voicemail :) |
| 01:08.19 | drmessano | No Alison |
| 01:08.21 | drmessano | :) |
| 01:08.58 | drmessano | Wonder if I can fax |
| 01:10.46 | drmessano | Apparently its *possible*, guess we'll need to see if Asterisk's implementation works well for it |
| 01:11.12 | leifmadsen | well, it's probably more a network connectivity thing |
| 01:13.24 | *** join/#asterisk trebaum (n=trebaum@ip68-8-175-208.sd.sd.cox.net) |
| 01:13.32 | drmessano | Skype for asterisk << Fax for Asterisk << T38 passthru << SPA3102 << RCA wireless phone extender << $59 HP Fax machine === WIN |
| 01:14.11 | drmessano | If that shit doesnt work, I want Mark Spencer at my house troubleshooting.. ASAP |
| 01:14.58 | raden_work | I have a phone that when i dial out the second someone picks up the call is dropped |
| 01:15.21 | drmessano | raden_work: CIA |
| 01:15.28 | raden_work | ? |
| 01:15.49 | drmessano | Sounds like codec negotiation or NAT |
| 01:15.59 | raden_work | well all other phones work |
| 01:16.08 | raden_work | this is the one wifi phone we have |
| 01:16.10 | drmessano | Im sure they do |
| 01:16.18 | drmessano | Check the config |
| 01:16.35 | drmessano | Is it the one on the daisychained Linksys and Netgear routers? |
| 01:17.11 | drmessano | Where when you plug the linksys wan port into a lan port on the netgear youve added another NAT? |
| 01:17.17 | raden_work | yeah the wifi router is now part of the lan though |
| 01:17.23 | raden_work | it shouldnt be doing any NAT |
| 01:17.30 | raden_work | netgear handing all DHCP |
| 01:17.51 | drmessano | Linksys LAN port is connected to the netgear? |
| 01:17.53 | raden_work | im running DDWRT plugged them lan to lan just to be sure |
| 01:18.13 | raden_work | also have wan in ddwrt to be set as part of switch as lan |
| 01:18.17 | raden_work | drmessano, yes sir |
| 01:19.00 | drmessano | I would still check the phone config |
| 01:19.07 | raden_work | call in office between all phones just fine |
| 01:19.11 | raden_work | i can dial out fine |
| 01:19.11 | drmessano | make sure you dont have some codec forced youre not supporting |
| 01:19.17 | raden_work | CLI not telling me anything |
| 01:19.27 | raden_work | ulaw |
| 01:19.31 | henry_ | Has anyone tried the Auto-dial out and deliver a prerecorded message example? |
| 01:19.36 | drmessano | The wifi phone can call other phones? |
| 01:19.48 | raden_work | <PROTECTED> |
| 01:19.49 | raden_work | <PROTECTED> |
| 01:19.49 | raden_work | <PROTECTED> |
| 01:19.51 | raden_work | thats all i get |
| 01:20.01 | drmessano | The wifi phone can call other phones just fine |
| 01:20.03 | drmessano | ? |
| 01:20.03 | raden_work | drmessano, yes any internal phone just fine |
| 01:20.13 | drmessano | canreinvite=no |
| 01:20.30 | raden_work | already seyt |
| 01:20.48 | drmessano | Perhaps the phone is telling you lies.. |
| 01:20.53 | raden_work | i can ring my cell but second i pickup call drops |
| 01:20.59 | drmessano | factory reset.. Default should not do that |
| 01:21.40 | raden_work | its a quickphones qa-342 |
| 01:21.52 | raden_work | it could be network issue but doubting it |
| 01:22.03 | raden_work | ill hook up a phone to lan on the linksys once and try |
| 01:22.23 | drmessano | That wont tell you anything |
| 01:23.42 | raden_work | well if there a issue on the router it will |
| 01:24.32 | raden_work | SPI firewall was active on lan got it |
| 01:24.34 | raden_work | disabled it |
| 01:24.41 | raden_work | everything goes through netgear anyway |
| 01:24.44 | raden_work | thanks for help |
| 01:25.22 | raden_work | omfg ok didnt get it |
| 01:25.54 | raden_work | and i only have one way audio out nothing in |
| 01:26.37 | drmessano | Sounds like you have some sort of NAT |
| 01:27.00 | drmessano | You need to recheck your configs |
| 01:28.45 | raden_work | should gateway address on phone be actual net gateway or the asterisk box ? |
| 01:28.50 | drmessano | Take the linksys, reflash with the linksys firmware, turn off DHCP, change the IP address, plug port 4 of the linksys into any port on the netgear, superglue the remaining ports on the Linksys |
| 01:29.32 | drmessano | Dunno.. gateway tells me nothing.. take it in context with the section of the config its in |
| 01:30.07 | drmessano | If its under NETWORK, then its the default gateway for the LAN, if its under the SIP SETTINGS, it probably doesnt mean the same |
| 01:30.43 | raden_work | i doubt its the firmware on the linksys |
| 01:31.24 | drmessano | You have doubted a lot of things, but it sounds like you either (A) Dont know how to config DD-WRT or (B) Dont know how to configure the phone |
| 01:31.51 | drmessano | Lowest common denominator.. if you want the linksys to be a dumb AP, make it DUMB |
| 01:32.37 | *** join/#asterisk BeeBuu (n=beebuu@61.145.77.6) |
| 01:32.57 | geneticx | Hello all. is version 8.3 the latest cisco 7960 ip phone software release? |
| 01:33.35 | *** join/#asterisk ingenius (n=alektro@vpn.itshidden.com) |
| 01:33.53 | geneticx | *8.5 i meant |
| 01:34.41 | raden_work | i only have 1 way audio on internal network as well |
| 01:35.52 | drmessano | Reset the DD-WRT box, turn off DHCP, change the IP, and plug it in the LAN.. leave the bridging crap alone |
| 01:35.59 | drmessano | It SHOULD only work as an AP |
| 01:37.08 | raden_work | its as a AP |
| 01:37.13 | raden_work | DHCP is disabled |
| 01:37.21 | geneticx | someone can confirm? |
| 01:39.59 | raden_work | ill disable dynamic routing |
| 01:41.14 | box2 | i have to disable my whole dialplan and start from scratch |
| 01:41.15 | box2 | gawd |
| 01:41.20 | box2 | breaks things |
| 01:43.44 | raden_work | this is ridiculous |
| 01:43.55 | shido6 | :) |
| 01:43.57 | raden_work | this phone has crap for settings |
| 01:45.08 | raden_work | OMFG pinging the wifi phone on the network i have 62 ms latency |
| 01:45.36 | raden_work | i dont understand that |
| 01:46.08 | drmessano | raden_work: You can insist all you want.. Your DD-WRT box is NOT just acting as a dumb AP |
| 01:46.09 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
| 01:46.39 | drmessano | You've obviously added some firewall component between the wireless and the LAN |
| 01:47.27 | raden_work | :( |
| 01:47.38 | raden_work | blah |
| 01:48.06 | drmessano | [21:36] <drmessano> Reset the DD-WRT box, turn off DHCP, change the IP, and plug it in the LAN.. leave the bridging crap alone |
| 01:49.10 | raden_work | already did that |
| 01:50.15 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 01:51.15 | raden_work | i wish i had my buffalo wrt-54g here never have a issue with that |
| 01:51.31 | drmessano | buffalo doesnt make the WRT-54G |
| 01:51.54 | raden_work | who does ? |
| 01:51.59 | raden_work | gah |
| 01:52.00 | drmessano | Linksys |
| 01:52.03 | raden_work | i meant my other one |
| 01:52.09 | raden_work | my aior station |
| 01:52.18 | drmessano | Have you been drinking? |
| 01:52.29 | drmessano | Maybe you should try this sober |
| 01:52.46 | raden_work | WHR-HP-54G |
| 01:52.54 | raden_work | lol no just tired |
| 01:53.19 | raden_work | freaking BOSS up my a i told hime give me a month so i can have time todo this first time setting up asterisk he gives me 6 days |
| 01:53.23 | raden_work | and 20 phones |
| 01:53.25 | raden_work | and yeah |
| 01:55.07 | raden_work | this is just not working |
| 01:55.42 | manxpower | It looks like you have an impossible task without the proper resources. Time to send out your resume. |
| 01:56.56 | raden_work | everything is working just not these stupid wifi phones :( |
| 01:57.09 | raden_work | and of course god forbid we buy anything decent |
| 01:58.11 | raden_work | why would i be getting audio out but not in on internal phones |
| 01:58.35 | drmessano | Because internal refers to the roof the walls, not the network in this case |
| 01:58.54 | drmessano | *and |
| 01:59.09 | raden_work | why are my pings from 19-72 ms |
| 01:59.17 | raden_work | nothing is steady as far as pings go |
| 01:59.38 | manxpower | I've never heard of a decent wifi phone |
| 01:59.51 | drmessano | Your network is a big NAT wirenut, thats why |
| 02:00.01 | drmessano | twist the wires, smell |
| 02:00.06 | *** part/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 02:00.16 | raden_work | I have i device for NAT |
| 02:00.17 | raden_work | thats it |
| 02:00.59 | drmessano | If youre so convinced this is working as you think, put everything behind the Linksys, set it up to be the ONLY device and try the wifi phone |
| 02:02.05 | raden_work | its something with that linksys |
| 02:02.21 | raden_work | it has to still be doing nat even though its not |
| 02:02.26 | raden_work | linksys firmware same crap |
| 02:03.12 | raden_work | Operating Mode |
| 02:03.12 | raden_work | <PROTECTED> |
| 02:03.18 | raden_work | i have it set to router |
| 02:03.32 | raden_work | so its not suppose to be doing NAT |
| 02:03.45 | *** join/#asterisk arthax0r (i=arthax0r@gateway/shell/blinkenshell.org/x-pjootdusnazcafnf) |
| 02:03.50 | drmessano | [22:02] <raden_work> its something with that linksys |
| 02:03.51 | drmessano | [22:02] <raden_work> it has to still be doing nat even though its not |
| 02:03.59 | drmessano | ^^^^^^^^^^^^^^^^^^^^^^^ Been saying that for an HOUR now |
| 02:04.07 | raden_work | thats how its acting but its disabled :( |
| 02:04.09 | drmessano | ~cluebat |
| 02:04.10 | infobot | *WHACK* *WHACK* *WHACK* |
| 02:04.16 | raden_work | lol |
| 02:04.31 | drmessano | I dont give a shit what your checkboxes tell you, man.. use some common sense |
| 02:04.36 | raden_work | everything is sitting on 192.168.1.0 network |
| 02:04.44 | raden_work | drmessano, wtf should i do |
| 02:04.48 | drmessano | <PROTECTED> |
| 02:04.53 | raden_work | ddwrt nor linksys firmware made a diffrence |
| 02:05.17 | drmessano | Linksys firmware, plug PORT 4 of the Linksys SWITCH into the NETGEAR |
| 02:05.26 | drmessano | NOT THE WAN PORT |
| 02:05.46 | raden_work | that the way i had it the whole time |
| 02:05.49 | raden_work | LAN TO LAN |
| 02:05.59 | raden_work | PORT 4 to PORT 8 |
| 02:06.02 | drmessano | With DD-WRT |
| 02:06.05 | raden_work | YES |
| 02:06.08 | drmessano | FAIL |
| 02:06.10 | drmessano | Go back |
| 02:06.13 | raden_work | ad linksys |
| 02:06.17 | raden_work | go back to ? |
| 02:06.18 | drmessano | .... |
| 02:06.37 | ricko73 | drmessano: common sense is not all that common |
| 02:06.58 | drmessano | The Linksys firmware DOES NOT and WILL NOT put a layer between the wireless and LAN ports.. |
| 02:07.24 | raden_work | i DMZ'ed the freakin IP phone still same crap |
| 02:07.26 | drmessano | I've resused Stock Linksys routers like that quite a bit |
| 02:07.31 | raden_work | linksys firmware same crap |
| 02:07.33 | drmessano | Stop guessing |
| 02:07.47 | drmessano | You flashed it back that quick? |
| 02:07.55 | raden_work | takes me like 45 seconds |
| 02:08.11 | raden_work | i've flashed a good 200 routers this year becoming second nature |
| 02:08.13 | drmessano | It takes that long to load the firmware, and another 2 mins to reboot completely |
| 02:08.21 | ricko73 | **POW BAM ZAP** zoom zoom batman |
| 02:08.43 | drmessano | raden_work: You flashed 200, and you sound like you can barely configure the ONE you have |
| 02:08.58 | drmessano | and second nature doesnt make the LOAD process take 45 secs |
| 02:09.02 | raden_work | never had a issue configuring anything till this dam ip phone |
| 02:09.05 | drmessano | You need a flux capacitor for that |
| 02:09.32 | raden_work | maybe i should flash dd-wrt voip on here |
| 02:09.37 | ricko73 | damn IP phone. dam is a physical structure usually used to hold back water |
| 02:09.39 | drmessano | That wont do shit |
| 02:09.48 | drmessano | You dont need a SIP proxy |
| 02:09.59 | raden_work | what should i do ? |
| 02:10.04 | drmessano | I already told you |
| 02:10.08 | raden_work | why are my ping times so irative |
| 02:10.23 | raden_work | erratic |
| 02:10.24 | drmessano | If you would stop stretching the truth here and do something I tell you, maybe you could get this going |
| 02:10.33 | drmessano | Put the god damn Linksys firmware back on |
| 02:10.44 | drmessano | Change the IP, turn DHCP off |
| 02:10.51 | raden_work | fine ill flash back to linksys firmware for the 5th time today |
| 02:11.01 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 02:12.38 | *** join/#asterisk Rob3Rt (i=Rob3Rt@181.45.96.58.static.exetel.com.au) |
| 02:12.41 | Rob3Rt | hi |
| 02:12.51 | Rob3Rt | anyone got a linksys wrt54gp2 here? |
| 02:13.01 | Rob3Rt | got crackly, troublesome voip issues |
| 02:13.14 | Rob3Rt | using g729 with nice bandwidtht |
| 02:13.45 | raden_work | hmmm |
| 02:14.06 | raden_work | update firmware |
| 02:14.12 | raden_work | very comon problem with that model |
| 02:15.24 | drmessano | What, that they come out with new firmware? |
| 02:15.24 | drmessano | I hear that a lot of routers have that problem |
| 02:15.40 | Rob3Rt | hmm |
| 02:15.55 | raden_work | linksys is shit |
| 02:15.56 | Rob3Rt | checking for an update, but just updated a few months back |
| 02:15.58 | Rob3Rt | yea |
| 02:16.21 | raden_work | my buffalo's with ddwrt work 10 x better |
| 02:16.29 | raden_work | my elcheapo asus even rocks |
| 02:16.53 | raden_work | what are the odds of bricking this thing again today |
| 02:20.36 | raden_work | LOL linksys back on same crap |
| 02:20.47 | Rob3Rt | stupid linksys |
| 02:21.04 | Rob3Rt | website doesnt know where its firmware is |
| 02:21.08 | Rob3Rt | linksys.com.au |
| 02:21.12 | Rob3Rt | stupid linksys - hate it |
| 02:21.25 | Rob3Rt | and it cant run third party |
| 02:21.30 | Rob3Rt | pityful :( |
| 02:22.25 | [TK]D-Fender | This from the person running AsteriskWin <- |
| 02:22.44 | *** join/#asterisk ArchGT (n=archgt@unaffiliated/archgt) |
| 02:26.00 | drmessano | <PROTECTED> |
| 02:26.04 | *** join/#asterisk Rob3Rt (n=admin@181.45.96.58.static.exetel.com.au) |
| 02:26.07 | Rob3Rt | sup |
| 02:26.13 | Rob3Rt | stupid linkshitz. |
| 02:26.25 | drmessano | I flashed 20+ Linksys WRT54G's and never had a problem with them on DD-WRT |
| 02:26.55 | *** part/#asterisk ArchGT (n=archgt@unaffiliated/archgt) |
| 02:26.56 | *** join/#asterisk Iamnacho (i=Iamnacho@98.186.180.143) |
| 02:27.03 | *** join/#asterisk ArchGT (n=archgt@unaffiliated/archgt) |
| 02:27.13 | KavanS | openwrt? |
| 02:27.17 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 02:27.31 | drmessano | DD-WRT |
| 02:28.00 | drmessano | Every now and then run into some little DD-WRT bug, but nothing Linksys specific |
| 02:28.52 | raden_work | this is bs with linksys firmware i cant even freakin ping ip phone |
| 02:29.52 | drmessano | raden_work: Are you like 19 or something? |
| 02:30.06 | raden_work | why you say stuff like that |
| 02:30.30 | raden_work | im frusterated is what i am this is ridiculous all i need todo is add a AP to our existing network which is not working |
| 02:30.37 | drmessano | Because your troubleshooting is "Stab, stab, reboot, stab, bitch, stab, reboot, reboot, reboot, reboot, stab" |
| 02:31.16 | drmessano | Its just a router.. Set it up like I told you and it will work fine |
| 02:31.58 | box2 | "how many times did you restart your computer?" "three times dude, just like you told me." |
| 02:32.02 | drmessano | Its like Network 101 feat. Holden Caulfield |
| 02:32.21 | raden_work | 64 bytes from 192.168.1.120: icmp_seq=3 ttl=255 time=38.7 ms |
| 02:32.21 | raden_work | 64 bytes from 192.168.1.120: icmp_seq=4 ttl=255 time=58.3 ms |
| 02:32.22 | raden_work | 64 bytes from 192.168.1.120: icmp_seq=5 ttl=255 time=84.3 ms |
| 02:32.22 | raden_work | 64 bytes from 192.168.1.120: icmp_seq=6 ttl=255 time=104 ms |
| 02:32.26 | raden_work | it got worse with linksys |
| 02:33.33 | drmessano | Are you sure your Vista box isnt the latency? |
| 02:33.46 | *** join/#asterisk lost_soul (i=shawn@cpe-74-71-234-29.twcny.res.rr.com) |
| 02:34.04 | raden_work | drmessano, ok listen very carefully, I have tried both DD-WRT and Linksys firmware ok The LAN port #4 on the WRT54GL is connected to the netgear, DHCP on linksys is disabled firewall disabled everything is disabled |
| 02:34.21 | raden_work | drmessano, there is not even a freakin windows box here |
| 02:34.34 | raden_work | do people just like to annoy people |
| 02:35.16 | drmessano | raden_work: If you disabled anything other than DHCP, you failed to follow my directions.. I do NOT trust your ability to check boxes or toggle radio buttons properly, and I was VERY specific with my instructions |
| 02:35.31 | drmessano | Do people just not like to listen to shit? |
| 02:35.49 | Rob3Rt | drmessano, which router is it ? |
| 02:35.50 | Rob3Rt | oh wait |
| 02:35.57 | drmessano | Youre fucking click happy, and its pissing me off |
| 02:36.10 | raden_work | <drmessano> Linksys firmware, plug PORT 4 of the Linksys SWITCH into the NETGEAR |
| 02:36.12 | Rob3Rt | yeah, that version doesnt support wireless bridging |
| 02:36.20 | Rob3Rt | lul |
| 02:36.25 | drmessano | Dont need bridging |
| 02:36.28 | Rob3Rt | k |
| 02:36.32 | raden_work | i reset factory defaults and started from there |
| 02:36.37 | Rob3Rt | ok |
| 02:36.50 | Rob3Rt | put the router on same subnet ? |
| 02:36.57 | box2 | i reset my linux to torvalds defaults |
| 02:36.57 | Rob3Rt | changed lan ip to suit ? |
| 02:37.02 | drmessano | Stock linksys firmware, factory reset, port 4 of the linksys in the other router, DHCP off, IP set to something non-conflicting, should work as a dumb AP |
| 02:37.04 | *** join/#asterisk LiNeTuX (n=LiNeTuX@119.153.205.68.cfl.res.rr.com) |
| 02:37.04 | Rob3Rt | ahh torvalds defaults |
| 02:37.07 | raden_work | netgear 192.168.1.1 linksys 192.168.1.2 |
| 02:37.18 | raden_work | conected via lan ports 4 & 8 |
| 02:37.19 | Rob3Rt | kewl |
| 02:37.32 | drmessano | But Holden Caulfield over there cant follow directions |
| 02:37.33 | raden_work | drmessano, thats what i have setup bro |
| 02:37.36 | *** join/#asterisk blkry (n=chatzill@96.37.27.72) |
| 02:37.41 | raden_work | thats what ive been trying to say |
| 02:37.45 | Rob3Rt | ic |
| 02:37.46 | raden_work | and my latency is ridiculous |
| 02:37.53 | raden_work | and i only have 1 way audio still |
| 02:37.54 | trebaum | raden_work you sure it's not a bad cable? |
| 02:37.54 | Rob3Rt | icanhazcheeseburger |
| 02:38.04 | raden_work | trebaum, changed it already :( |
| 02:38.10 | trebaum | ok. |
| 02:38.11 | Rob3Rt | lululululululululululululululu fried |
| 02:38.13 | raden_work | ping to router is less than 1 ms |
| 02:38.19 | box2 | icanhazping |
| 02:38.21 | Rob3Rt | sure solution - remove linksys from network |
| 02:38.29 | raden_work | yeah i agree |
| 02:38.32 | Rob3Rt | add in something ... else |
| 02:38.37 | Rob3Rt | even if you have to, |
| 02:38.40 | henry_ | I am having an issue creating multiple calls using call files and playing a message. The first call issued works fine, the others just seem to hang up after they are answered. Has anyone seen a similar problem? |
| 02:38.48 | raden_work | drmessano, im not trying to be dumb ive been trying to look at this with logic for 4 hours |
| 02:38.54 | box2 | nothing wrong with linksys wrt54g routers |
| 02:38.55 | Rob3Rt | draw a picture of a router, and sticky tape it to the network, it would more than likely be better than the linksys |
| 02:39.04 | Rob3Rt | lies |
| 02:39.09 | Rob3Rt | my wrt54gp2 is shiz |
| 02:39.10 | raden_work | Rob3Rt, thanks for the laugh i needed that |
| 02:39.11 | drmessano | box2: Exactly.. Not a $1000 router, but they do well |
| 02:39.19 | Rob3Rt | raden_work, no probs lol :) |
| 02:39.38 | raden_work | drmessano, PING 192.168.1.120 (192.168.1.120) 56(84) bytes of data. |
| 02:39.38 | raden_work | 64 bytes from 192.168.1.120: icmp_seq=1 ttl=255 time=95.1 ms |
| 02:39.38 | raden_work | 64 bytes from 192.168.1.120: icmp_seq=2 ttl=255 time=209 ms |
| 02:39.39 | *** join/#asterisk OrNix (n=ornix@78.40.81.34) |
| 02:39.48 | drmessano | Ping the LINKSYS |
| 02:39.50 | raden_work | PING 192.168.1.2 (192.168.1.2) 56(84) bytes of data. |
| 02:39.50 | raden_work | 64 bytes from 192.168.1.2: icmp_seq=1 ttl=64 time=0.541 ms |
| 02:39.50 | raden_work | 64 bytes from 192.168.1.2: icmp_seq=2 ttl=64 time=0.524 ms |
| 02:39.57 | Rob3Rt | kewl |
| 02:39.57 | drmessano | ok |
| 02:40.01 | raden_work | 120 ip phone |
| 02:40.01 | Rob3Rt | ping google.com.au |
| 02:40.04 | Rob3Rt | isee |
| 02:40.07 | Rob3Rt | fixexd? |
| 02:40.34 | drmessano | Maybe the phone is a piece of shit.. |
| 02:40.35 | raden_work | ping www.google.com.au |
| 02:40.35 | raden_work | PING www.l.google.com (209.85.225.99) 56(84) bytes of data. |
| 02:40.35 | raden_work | 64 bytes from iy-in-f99.google.com (209.85.225.99): icmp_seq=1 ttl=55 time=56.6 ms |
| 02:40.35 | raden_work | 64 bytes from iy-in-f99.google.com (209.85.225.99): icmp_seq=2 ttl=55 time=62.9 ms |
| 02:40.40 | Rob3Rt | yay |
| 02:40.42 | Rob3Rt | sorted |
| 02:40.43 | drmessano | Isnt it a diahatsu |
| 02:40.44 | raden_work | drmessano, i was starting to think that |
| 02:41.00 | Rob3Rt | prolly cable, make sure theyre all plugged right in |
| 02:41.01 | drmessano | ~happyclownpbx |
| 02:41.02 | infobot | [HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for it's core, it pwns, is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone |
| 02:41.12 | Rob3Rt | thats usually ot |
| 02:41.15 | Rob3Rt | it* |
| 02:41.15 | drmessano | ~Diahatsumashiniriki Keyotason 200LP-A11 |
| 02:41.18 | drmessano | hmmm |
| 02:41.21 | raden_work | huh |
| 02:41.22 | drmessano | ~Diahatsumashiniriki Keyotason |
| 02:41.28 | drmessano | GAH |
| 02:41.32 | drmessano | ~Diahatsumashiniriki |
| 02:41.35 | drmessano | :( |
| 02:41.41 | drmessano | I had a trigger in there somewhere |
| 02:42.06 | drmessano | Diahatsumashiniriki Keyotason 200LP-A11 SIP phone <-- Your typical shit eBay special |
| 02:42.18 | Rob3Rt | spa-942 ftw |
| 02:42.23 | drmessano | $29 Supper Happy SIP phone |
| 02:42.25 | raden_work | ping 192.168.1.3 |
| 02:42.26 | raden_work | PING 192.168.1.3 (192.168.1.3) 56(84) bytes of data. |
| 02:42.26 | raden_work | From 192.168.1.100: icmp_seq=1 Destination Host Unreachable |
| 02:42.26 | raden_work | From 192.168.1.100 icmp_seq=1 Destination Host Unreachable |
| 02:42.29 | Rob3Rt | lol |
| 02:42.32 | Rob3Rt | hm |
| 02:42.33 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 02:42.34 | raden_work | thats my laptop on same wifi |
| 02:42.39 | Rob3Rt | hm |
| 02:42.43 | Rob3Rt | firewalled ? |
| 02:42.48 | raden_work | nope |
| 02:42.49 | Rob3Rt | to drop icmp perhaps |
| 02:42.52 | blkry | anybody used the new cisco SPA504G? |
| 02:42.53 | Rob3Rt | hm |
| 02:42.56 | raden_work | that could |
| 02:43.05 | box2 | did you check your route settings |
| 02:43.52 | raden_work | thats windows laptop :( |
| 02:43.57 | *** join/#asterisk alrs (n=lars@rrcs-24-43-46-25.west.biz.rr.com) |
| 02:44.02 | drmessano | HA |
| 02:44.23 | raden_work | ? |
| 02:44.26 | drmessano | [22:34] <raden_work> drmessano, there is not even a freakin windows box here |
| 02:44.37 | drmessano | Youre a fuckwit |
| 02:44.41 | drmessano | No, really |
| 02:44.43 | raden_work | there wasnt i went out to service van and got laptop but yes there is one here now |
| 02:44.51 | drmessano | In 18 seconds? |
| 02:44.53 | raden_work | its all opensuse on everything though |
| 02:45.02 | raden_work | service van behind building jesus |
| 02:45.04 | LiNeTuX | ahhh, THAT Windows |
| 02:45.40 | raden_work | whats average latency over wifi with a wifi phone ? |
| 02:45.55 | Rob3Rt | 5-10ms i would imagine |
| 02:46.11 | LiNeTuX | what is the airspeed of an unladen african swallow |
| 02:46.26 | drmessano | There hasnt been a span of more than 30 seconds you havent entered text in here.. considering average time to type a sentence, you must be sitting in that van and have an 8 core Vista laptop in hybernation with a Flash disk |
| 02:46.34 | Rob3Rt | err i get 1ms over wireless via a 3 point network |
| 02:47.01 | [TK]D-Fender | LiNeTuX: I don't know... |
| 02:47.05 | drmessano | Al lies.. youre probably setting up Skype for your mom.. not asterisk for your boss |
| 02:47.07 | [TK]D-Fender | AAAAAAAAAAAARRRRRRRRGGGGGGGGGGGGGGGGGHHHHHHHHHHHHHHH!!!!!!!!!!!!!!!!!! |
| 02:47.13 | Rob3Rt | LiNeTuX, the speed is variable. |
| 02:47.30 | [TK]D-Fender | Rob3Rt: pop culture FAIL |
| 02:47.38 | drmessano | LiNeTuX: You ruined the joke |
| 02:47.51 | drmessano | Unladen swallow.. |
| 02:47.52 | [TK]D-Fender | drmessano: I saved it by being first |
| 02:47.56 | Rob3Rt | if you asked, what is the speed of a swallow flying at 10kms per hour, i could give a difinitive answer |
| 02:48.05 | Rob3Rt | [TK]D-Fender, i hate pop |
| 02:48.09 | drmessano | Unladen african swallow.. you ruined the punchline |
| 02:48.20 | LiNeTuX | yeah yeah yeah |
| 02:48.22 | Rob3Rt | whats the punchline ? |
| 02:48.25 | drmessano | Now I have to ask "North african or west african" or throw a color in there |
| 02:48.28 | [TK]D-Fender | Rob3Rt: My yogurt has more culture that you, and thats just the BACTERIA |
| 02:48.34 | drmessano | HAHAH |
| 02:48.54 | Rob3Rt | [TK]D-Fender, im still wondering what u think i did to fail lol |
| 02:48.57 | Rob3Rt | didnt say anything |
| 02:49.02 | drmessano | You're no Tim the Enchanter |
| 02:49.18 | [TK]D-Fender | LiNeTuX: And you weren't supposed to mention "african" |
| 02:49.47 | drmessano | I am an ennnchanter! |
| 02:49.54 | drmessano | There are some who call me.... Tim |
| 02:49.59 | LiNeTuX | [TK]D-Fender: I usually don't play by the rules, either. |
| 02:50.30 | drmessano | I called one of the guys at work "Sir Robin" the other day |
| 02:50.35 | drmessano | I got some laffs.. it was worth it |
| 02:51.42 | LiNeTuX | had a good day today. The first time *ever* setting up an Asterisk box - getting everything done before the PRI got provisioned - provider gets their stuff done, I plug it in, and it works w/o issue. |
| 02:52.21 | *** join/#asterisk voxter (n=voxter@S0106002369b3cd56.vc.shawcable.net) |
| 02:53.21 | drmessano | LiNeTuX: Badass |
| 02:53.47 | raden_work | drmessano, since your so smart why is this not working ? |
| 02:53.52 | LiNeTuX | drmessano: not really. but it was a nice change from the usual back-and-forth troubleshooting w/the provider. |
| 02:54.19 | LiNeTuX | It probably helped that the CO was only 30' away. |
| 02:54.53 | drmessano | LiNeTuX: Sure it was.. Take what you can get, man.. Install a workstation for your Dad and he doesnt bitch about his favorites for PornKing.com missing = Badass.. Install an * box and no users are hurt in he process = Badass |
| 02:55.09 | LiNeTuX | heh |
| 02:55.22 | LiNeTuX | "Everyone stand back! I'm going to PLUG IT IN!" |
| 02:55.23 | drmessano | raden_work: We've been over this time and time again |
| 02:55.46 | raden_work | well i did exactly as you said |
| 02:56.28 | raden_work | took linksys router with defaults, disabled DHCP, set a static non conflicting ip, plugged port #4 into netgear lan and it still dont work |
| 02:56.31 | raden_work | what did i miss ? |
| 02:56.50 | drmessano | http://tinyurl.com/5zx54h <-- Follow that |
| 02:57.01 | raden_work | and then i switched it from gateway to router which shouldnt matter but according to DDwrt and Linksys docs on both it totally disables NAT |
| 02:57.21 | drmessano | You dont need to disable ANYTHING |
| 02:57.30 | drmessano | Except DHCP |
| 02:57.35 | drmessano | Firewall, NO |
| 02:57.45 | raden_work | I did that the first time |
| 02:57.45 | drmessano | gateway to router, NO |
| 02:58.00 | raden_work | i didnt do crap besides set a static IP and disable DHCP |
| 02:58.04 | raden_work | it DONT WORK |
| 02:58.14 | raden_work | the phone still does not recieve AUDIO |
| 02:58.27 | raden_work | and my latency is almost 300 ms at times with linksys |
| 02:58.34 | raden_work | ddwrt never went over 80 i dont get that |
| 02:59.46 | drmessano | Im not sure what to believe.. you tell me youve done THIS, THIS, and THAT, then you tell me you did none of those things, only blah.. If you want this to work, i've told you 5 or 6 times how to set the Linksys up appropriately.. Maybe when youre done with whatever it is youre doing, you can do that |
| 03:00.14 | raden_work | OMFG |
| 03:00.29 | raden_work | im not a fing moron so stop treating me like one |
| 03:00.33 | raden_work | IT DONT WORK |
| 03:00.40 | drmessano | http://www.urbandictionary.com/define.php?term=click-happy |
| 03:00.46 | LiNeTuX | everyone's a moron at some point in their life. |
| 03:00.54 | drmessano | I never said you are a moron, i said you're click happy |
| 03:01.03 | raden_work | im not click happy either |
| 03:01.13 | LiNeTuX | has a click-happy 3.5 year old |
| 03:01.15 | raden_work | this is ridiculous im having NAT like issues with no nat |
| 03:01.49 | raden_work | and all you want to do is sit here ad make fun of this stressful situation |
| 03:02.04 | LiNeTuX | raden: did you try a softphone? |
| 03:02.28 | drmessano | all you want to do is waste our time |
| 03:02.35 | drmessano | Tried to help you for hours now |
| 03:03.00 | drmessano | If you want, I can play dumbass for you and pretend the Linksys doesn't work as described |
| 03:03.02 | [TK]D-Fender | [23:00]<LiNeTuX>everyone's a moron at some point in their life. <- some seem Hell-bent on making that moment last a LIFETIME |
| 03:03.06 | shido6 | oh come on now people :) |
| 03:03.36 | shido6 | NAT issues without NAT sounds pretty frustrating |
| 03:03.51 | drmessano | http://forums.techguy.org/networking/476997-wrt54g-access-point-only.html |
| 03:04.02 | drmessano | shido6: So does asking someone a dozen times to reset a fuckin router |
| 03:04.21 | drmessano | shido6: When do you get to the point of releasing the hounds |
| 03:05.32 | drmessano | A: About 2 hours ago |
| 03:05.50 | raden_work | wow thats exactly what |
| 03:05.53 | raden_work | I DID |
| 03:06.00 | drmessano | 20-20-20 |
| 03:06.09 | raden_work | everything is working as far as the router and wifi and routing |
| 03:06.20 | raden_work | but the freaking WIFI phone only has 1 way audio |
| 03:06.22 | drmessano | Except its not |
| 03:06.55 | drmessano | Its not at all working how you say, you insist it is based on what? That it PINGS? |
| 03:07.18 | carrar | ping is 100% |
| 03:07.24 | carrar | heh |
| 03:07.29 | raden_work | wow i can register the wifi phone i can dial out on it my wifi laptop connects fine dhcp addresses are assigned properly |
| 03:07.37 | carrar | PING FTW!! |
| 03:07.45 | drmessano | Ping apparently gives a total network snapshot |
| 03:08.04 | drmessano | Maybe I dont know the correct switch to get that under Vista home basic |
| 03:08.05 | shido6 | so maybe your wifi phone cant deal with nat :) |
| 03:08.11 | raden_work | no but dont you find it odd my latency is almost 300 ms at times on an internal network ?> |
| 03:08.13 | shido6 | or maybe the firmware on the wifi phone cant deal with nat |
| 03:08.14 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-yzwauxkihylonvdw) |
| 03:08.15 | drmessano | shido6: What NAT? heh |
| 03:08.21 | shido6 | you said router |
| 03:08.24 | drmessano | This is LOCAL |
| 03:08.26 | shido6 | so I say nat |
| 03:08.42 | shido6 | where's asterisk in this mix? |
| 03:08.44 | carrar | I say potatoe! |
| 03:08.45 | raden_work | shido6, i only have 1 way audio when im calling phone to phone on lan as well |
| 03:08.53 | drmessano | WRT54G ---Wired as AP, LAN <> LAN ---> Netgear as gateway.. |
| 03:09.07 | drmessano | Asterisk box behind Netgear |
| 03:09.13 | drmessano | Well, on the same side |
| 03:09.17 | raden_work | yeah same side |
| 03:09.17 | shido6 | .... |
| 03:09.32 | drmessano | Wireless <> LAN on Linksys acts like theres a NAT/Firewall |
| 03:09.34 | raden_work | I dont understand it the phone is acting like its behind a nat :( |
| 03:09.42 | drmessano | No crap |
| 03:09.47 | drmessano | Do a 100-100-100 reset |
| 03:09.54 | raden_work | Disabled firewall |
| 03:10.07 | raden_work | drmessano, as is restart everytthing here ? |
| 03:10.07 | drmessano | Again, clicking something you shouldnt have |
| 03:10.21 | raden_work | drmessano, why would disabling firewall even matter ? |
| 03:10.25 | drmessano | Do you know what changing from GATEWAY to ROUTER changes in the Linksys? |
| 03:10.26 | raden_work | netgear takes care of that |
| 03:10.35 | drmessano | DO YOU? |
| 03:10.37 | raden_work | drfreeze, yes changes from nat to classic routing |
| 03:10.42 | drmessano | Do you know if it changes bridging? |
| 03:10.53 | drmessano | Do you know how exactly it accomplishes it? |
| 03:11.09 | drmessano | Really, it doesnt matter.. Its NEEDLESS and something extra in the mix |
| 03:11.37 | raden_work | technically gateway to router shouldnt matter |
| 03:11.45 | raden_work | cause it should only be for the wan port |
| 03:11.50 | raden_work | which we are not messing with |
| 03:11.50 | drmessano | Want me to take this WRT54GL I have in this box next to me, reset it, set it up as an AP, and connect for you? |
| 03:11.58 | raden_work | but i have seen weirder glitches in firmwre |
| 03:12.10 | raden_work | thats what i have |
| 03:12.12 | drmessano | raden_work: Technically, you have no idea what the router is doing, and youve checked something based on a hunch |
| 03:12.12 | carrar | drmessano, will you do that AND cook dinner? |
| 03:12.19 | raden_work | i can make it a access point |
| 03:12.35 | raden_work | drmessano, do you listen ? |
| 03:12.44 | raden_work | at all or are you just arrogant ? |
| 03:12.51 | drmessano | raden_work: You cant follow basic directions.. reset the fucking thing, 30-30-30, turn off DHCP, Set the IP, connect LAN <> LAN |
| 03:12.59 | raden_work | 30-30-30 ? |
| 03:13.00 | drmessano | You dont listen.. thats the problem |
| 03:13.08 | raden_work | ive been listening |
| 03:13.16 | raden_work | what does 30-30-30 mean ? |
| 03:13.27 | drmessano | 30 plugged in, holding, pull power for 30, still holding, plug back in, hold for 30 more |
| 03:13.29 | carrar | brast, hips, waste |
| 03:13.31 | carrar | err |
| 03:13.33 | carrar | waist |
| 03:13.40 | raden_work | lol |
| 03:13.41 | carrar | oh I can't spell tonight |
| 03:13.53 | drmessano | Thats a 30-30-30 reset |
| 03:13.53 | raden_work | drmessano, ok ill reset the dreaking router again |
| 03:14.09 | drmessano | 30-30-30, and I am counting |
| 03:14.18 | LiNeTuX | Wait, it's a dreaking? I thought it was a Linksys. |
| 03:14.50 | drmessano | dreaking is a unit of time where he can perform a 90 sec reset cycle in 18 seconds and have it already booted back up |
| 03:15.14 | LiNeTuX | oh, cool. like a time warp. |
| 03:15.19 | LiNeTuX | Let's do the time warp... again.... |
| 03:16.06 | LiNeTuX | It's just a port to the left.... and not nat to the right... |
| 03:16.08 | drmessano | raden horror picture show? |
| 03:16.31 | LiNeTuX | with your hands on your button |
| 03:16.35 | LiNeTuX | and the power out of sight |
| 03:16.43 | LiNeTuX | ok, i'm done |
| 03:18.03 | raden_work | drmessano, now what would u like me todo ? |
| 03:18.10 | [TK]D-Fender | drmessano: I prefer the 30-30 reset |
| 03:18.19 | [TK]D-Fender | drmessano: Or maybe the 30-06 reset more... |
| 03:18.27 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 03:18.33 | rob0 | But it's the pelvic thrusting, that really drives me in-sa-a-a-a-a-ane |
| 03:18.54 | [TK]D-Fender | uNF! uNF! uNF! uNF! uNF! uNF! |
| 03:19.03 | raden_work | is this just a big joke to everyone ? |
| 03:19.18 | LiNeTuX | He thinks it's about him. |
| 03:19.21 | LiNeTuX | How quaint. |
| 03:19.30 | raden_work | no the whole reset thing |
| 03:19.54 | raden_work | <PROTECTED> |
| 03:21.34 | drmessano | Already explained |
| 03:22.22 | raden_work | well its rebooted i disabled DHCP set IP to 192.168.1.2 |
| 03:22.27 | raden_work | now what |
| 03:22.48 | drmessano | Scroll up and read one of the 5 times I told you.. |
| 03:23.02 | raden_work | yeah did that now what ? |
| 03:23.11 | [TK]D-Fender | ~nowwhat |
| 03:23.12 | infobot | So you just installed Asterisk now what? http://www.youtube.com/watch?v=FJ3oHpup-pk |
| 03:23.28 | drmessano | well its rebooted i disabled DHCP set IP to 192.168.1.2 <-- That wasnt all I told you to do.. |
| 03:23.35 | drmessano | ~insanity |
| 03:23.35 | infobot | methinks insanity is good; if you're not a paranoiac, and the world really is out to get you, how will you know? |
| 03:23.44 | drmessano | gah |
| 03:23.49 | raden_work | drmessano, i rebooted after i did all that sorry |
| 03:24.07 | raden_work | wow people are worthless |
| 03:24.23 | drmessano | I was just thinking that |
| 03:24.24 | raden_work | i have 1 way freaking audio what is so difficult about this |
| 03:24.34 | drmessano | Its not |
| 03:24.48 | raden_work | i can plug a wired ip phone into the lan on the linksys and it works fine |
| 03:25.13 | raden_work | all the wireless phones have 1 way audio |
| 03:25.17 | alrs | fuhhhhhh |
| 03:25.19 | alrs | FREAK OUT! |
| 03:25.37 | shido6 | raden_work: so check firmware changelogs |
| 03:26.18 | raden_work | shido6, did that updated to newest firmware tried linksys, ddwrt, tomato firmware all sam issue |
| 03:26.24 | [TK]D-Fender | 1-way audio.... |
| 03:26.26 | [TK]D-Fender | \o/ |
| 03:26.27 | raden_work | yes sir |
| 03:26.37 | raden_work | i cant hear anything on wifi phone |
| 03:26.42 | raden_work | CLI not saying anything either |
| 03:26.45 | alrs | wifi network is same subnet as wired? |
| 03:26.53 | [TK]D-Fender | raden_work: because you're not looking |
| 03:27.18 | drmessano | raden_work: My final comment.. It sounds to me like you've got something set in NVRAM from your little DDWRT bridging experiment that your little bitch impatience could fix by clearing the NVRAM with a proper reset followed by a simple config of the linksys |
| 03:27.29 | [TK]D-Fender | drmessano: I wouldn |
| 03:27.37 | [TK]D-Fender | drmessano: I wouldn't even jump that far... |
| 03:27.49 | [TK]D-Fender | drmessano: we are so far ahead of the ball and nedlessly so |
| 03:27.52 | drfreeze | raden_work: any of this help: http://forum.pfsense.org/index.php?topic=504.msg%msg_id% |
| 03:27.55 | [TK]D-Fender | needlessly* |
| 03:29.05 | drmessano | Hes fighting a one way audio problem with WIRELESS devices only on a WRT54G he's configuring, supposedly, as an AP ONLY on his LAN, that he previously had DDWRT on (HELLLLO???) that he admittedly screwed with the bridging on... |
| 03:29.15 | drmessano | Thats really not a stretch, and easy to fix |
| 03:29.16 | raden_work | why does my aastra 9133i phones work with every firmware but the wifi phone will not ? |
| 03:29.29 | [TK]D-Fender | drmessano: Yeah, but we aren't seeing a single thing, and you know what that means... |
| 03:29.47 | [TK]D-Fender | raden_work: Because you are running around like a headless chicken and not looking |
| 03:29.57 | raden_work | cause everyone is argueing with me :( |
| 03:30.04 | drmessano | HAHA |
| 03:30.14 | [TK]D-Fender | raden_work: No, your inability to actually look at whats going on is YOUR fault. |
| 03:30.16 | drmessano | Everyone is telling you ----> DO THIS |
| 03:30.22 | raden_work | ive tried explaining my problem everyone keeps saying im screwing up the router |
| 03:30.33 | [TK]D-Fender | raden_work: You can ignore distractions and focus on something concrete |
| 03:30.39 | drmessano | and youre all liek ----> ZOMG UM TOOTHPICK HAMPSTER |
| 03:30.44 | [TK]D-Fender | raden_work: Stop explaining, and start looking. |
| 03:30.56 | raden_work | i have been for almost 6 hours |
| 03:31.11 | [TK]D-Fender | raden_work: And I haven't seen anyone even hint at the obvious. |
| 03:31.27 | drmessano | [TK]D-Fender: Factory reset trumps looking any day.. but then you have to assume clicking 3 boxes doesnt become "Oh like, 9 or so" |
| 03:31.40 | blkry | can you reformat the phone? Like a polycom? |
| 03:31.40 | drmessano | Obvious? |
| 03:31.45 | [TK]D-Fender | drmessano: Still just a story to me... |
| 03:31.53 | [TK]D-Fender | drmessano: Entirely. No-one is looking |
| 03:32.05 | [TK]D-Fender | GOD THE CRAZY PEOPLE |
| 03:32.12 | [TK]D-Fender | THEY'RE EVERYWHERE |
| 03:32.18 | drmessano | No one needs to look.. |
| 03:32.34 | raden_work | drmessano, i reset all factory defaults > turned DHCP OFF > SET STATIC IP > Rebooted router |
| 03:32.38 | drmessano | If you wipe and start over, you've removed the looking part |
| 03:32.44 | raden_work | same thing as when i had ddwrt setup |
| 03:32.46 | [TK]D-Fender | drmessano: No, they can just continue making abstract guesses and juicing up the BlameThrower |
| 03:33.14 | [TK]D-Fender | drmessano: Looking can happen at any point... any point that is, except for the last 6 HOURS |
| 03:33.21 | raden_work | as much as i have changed the router nothing has changed I know networking I dont know much about VOIP but i know networking |
| 03:33.47 | [TK]D-Fender | raden_work: Knows much, sees little. |
| 03:33.57 | blkry | looking or reconfigure. how long does it take to configure a phone? |
| 03:34.00 | [TK]D-Fender | raden_work: Can't ID what you aren't looking at |
| 03:34.02 | drmessano | So how exactly does looking at the questionable configuration of the device trump resetting it to a known unfucked state? |
| 03:34.20 | [TK]D-Fender | drmessano: We are guessing a state, thats the problem. We aren't looking at it. |
| 03:34.22 | raden_work | drmessano, im getting really sick of your attitude |
| 03:34.34 | drmessano | We dont need to guess the state if its reset |
| 03:34.37 | [TK]D-Fender | raden_work: So are you ready to actually look at the problem now? |
| 03:35.06 | raden_work | <PROTECTED> |
| 03:35.07 | drmessano | Apparently thats not so OBVIOUS |
| 03:35.15 | [TK]D-Fender | raden_work: .... |
| 03:35.19 | [TK]D-Fender | ~wmmfpb |
| 03:35.20 | infobot | [~wmmfpb] WHERE'S MY MOTHER-^%#ING PASTEBIN?!? |
| 03:35.21 | [TK]D-Fender | ???? |
| 03:35.38 | raden_work | would you like configs or debug output ? |
| 03:35.45 | [TK]D-Fender | Not one dimwit here seems to think looking at the DAMNED SIP COMMUNICATION might give something away |
| 03:35.50 | [TK]D-Fender | LOOKO AT THE FUCKING CALL! |
| 03:36.09 | [TK]D-Fender | Damn people... |
| 03:36.23 | [TK]D-Fender | All this guessing and noone is looking at the stupid 1-way audio CALL |
| 03:36.28 | [TK]D-Fender | Wake the hell up |
| 03:36.42 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 03:37.18 | raden_work | :( |
| 03:38.29 | [TK]D-Fender | "Think we should look at the body to determine how Jon died?" "Oh look, a kitten!" |
| 03:38.49 | [TK]D-Fender | interrogates the kitten |
| 03:39.13 | blkry | does it have rabies |
| 03:40.02 | raden_work | [TK]D-Fender, http://www.voltarclamps.com/files/sip.txt |
| 03:40.12 | [TK]D-Fender | unleashes the kitten on blkry and hopes for rabies |
| 03:40.17 | raden_work | lmao |
| 03:41.09 | [TK]D-Fender | raden_work: <--- Transmitting (NAT) to 192.168.1.120:5060 ---> <-- NAT, Pardon? |
| 03:41.51 | raden_work | how do i fix that ? |
| 03:42.22 | [TK]D-Fender | raden_work: Remote the net=yes from your peers. set canreinvite=no on them as well |
| 03:42.35 | raden_work | ok |
| 03:43.24 | raden_work | all of them ? |
| 03:43.46 | [TK]D-Fender | raden_work: Are any of them supposed to be behind a remote NAT? |
| 03:44.11 | raden_work | there all internal to the asterisk box |
| 03:44.26 | raden_work | there all on the same lan as asterisk box |
| 03:45.15 | raden_work | sip reload still 1 way audio |
| 03:46.39 | raden_work | [TK]D-Fender, ? |
| 03:46.57 | [TK]D-Fender | raden_work: I don't see an updated pastebin. |
| 03:50.53 | raden_work | [TK]D-Fender, http://www.voltarclamps.com/files/sip2.txt |
| 03:52.33 | raden_work | Sending to 192.168.1.120 : 5060 (NAT) ????????/ |
| 03:53.16 | raden_work | <-------------> |
| 03:53.16 | raden_work | --- (8 headers 0 lines) --- |
| 03:53.16 | raden_work | Sending to 192.168.1.120 : 5060 (NAT) |
| 03:53.40 | [TK]D-Fender | let me know when you can get it right |
| 03:53.55 | raden_work | get what right ? |
| 03:54.17 | [TK]D-Fender | [23:53]<raden_work>Sending to 192.168.1.120 : 5060 (NAT) <-- what do yout hink? |
| 03:54.31 | raden_work | i removed nat=yes from everything in sip.conf |
| 03:54.38 | raden_work | i just did a find and there is no nat in it |
| 03:55.26 | raden_work | did a sip and dialplan reload |
| 03:55.54 | [TK]D-Fender | did you say nat=no for your peers? |
| 03:56.16 | raden_work | [TK]D-Fender, you told me [TK]D-Fender> raden_work: Remote the net=yes from your peers. set canreinvite=no on them as well |
| 03:56.23 | raden_work | i removed them did not set anything to no |
| 03:56.29 | raden_work | should i set them to no ? |
| 03:56.44 | [TK]D-Fender | raden_work: Please try to be maybe even a LITTLE intuitive and define things explicitily |
| 03:57.18 | raden_work | i do that people tell me im click happy i follow things to the T and here I am again being told how i dont do things right :( |
| 03:57.40 | raden_work | feel like im walking on eggshells in here today |
| 03:58.03 | raden_work | hi everyone im retarted, stupid, cant read, follow instructions, have any common sense, so please treat me as so |
| 03:58.10 | raden_work | i get it |
| 04:00.08 | [TK]D-Fender | raden_work: new pastebin. |
| 04:00.16 | raden_work | ok |
| 04:00.23 | [TK]D-Fender | raden_work: and is 1.120 your WiFi Phone? |
| 04:00.31 | raden_work | yes sir |
| 04:00.39 | [TK]D-Fender | raden_work: Dump your firewall. |
| 04:00.51 | raden_work | they are |
| 04:01.00 | raden_work | disabled network wide open |
| 04:01.05 | [TK]D-Fender | raden_work: SHOW ME |
| 04:01.19 | raden_work | how you want a screen shot ? |
| 04:01.36 | raden_work | Netgear security set to allow all |
| 04:01.44 | raden_work | linksys SPI firewall = disabled |
| 04:01.51 | raden_work | everything has been restarted from there |
| 04:01.53 | [TK]D-Fender | ***IPTABLES*** |
| 04:03.49 | drmessano | Wired phones work |
| 04:04.00 | [TK]D-Fender | Don't care |
| 04:04.01 | drmessano | Its only the wireless ones on the Linksys |
| 04:04.03 | raden_work | [TK]D-Fender, flush or shutdown completly ? |
| 04:04.05 | shido6 | has your wifi phone EVER worked? |
| 04:04.15 | [TK]D-Fender | raden_work: Empty it. |
| 04:04.18 | raden_work | drmessano, the wired ones also work on the linksys just not the wireless |
| 04:04.42 | [TK]D-Fender | raden_work: and SHOW ME |
| 04:06.14 | raden_work | SERVER:/ # SuSEfirewall2 stop |
| 04:06.14 | raden_work | SuSEfirewall2: batch committing... |
| 04:06.14 | raden_work | SuSEfirewall2: Firewall rules unloaded. |
| 04:06.36 | raden_work | sorry used to debian couldnt find right away in /etc/init.d/ |
| 04:07.09 | raden_work | [TK]D-Fender, u are a genius |
| 04:07.17 | raden_work | so please explain whats the deal ? |
| 04:09.07 | raden_work | ok wtf |
| 04:09.23 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 04:09.23 | raden_work | i can call the 120 and have bi-directional but i still cant dial from it |
| 04:09.29 | raden_work | sorry that did not fix it |
| 04:09.34 | raden_work | another pastebin we go |
| 04:10.59 | [TK]D-Fender | raden_work: Sounds like it it fixed part of your issues |
| 04:11.14 | raden_work | i never tried calling the wifi phone before |
| 04:11.20 | raden_work | i just relized first time i did it |
| 04:12.26 | [TK]D-Fender | Or at least the problem you HAD. Now you have the opportunity to find all your OTHER ones. |
| 04:12.54 | raden_work | still cant here anything on WIFI phone |
| 04:12.57 | raden_work | same URL |
| 04:15.22 | raden_work | http://www.voltarclamps.com/files/sip2.txt |
| 04:15.56 | raden_work | --- (12 headers 12 lines) --- |
| 04:15.56 | raden_work | <PROTECTED> |
| 04:15.56 | raden_work | Sending to 192.168.1.120 : 5060 (NAT) |
| 04:15.56 | raden_work | why it still doing NAT ? |
| 04:17.28 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 04:17.49 | KyleK | because asterisk thinks it needs to? |
| 04:18.13 | carrar | because you never did "sip reload" ? |
| 04:18.55 | raden_work | every time |
| 04:19.35 | [TK]D-Fender | Why do I not see the updated CONFIGS? |
| 04:19.42 | [TK]D-Fender | Someone hasn't seemed to learn.... |
| 04:20.31 | *** part/#asterisk korihor (n=korihor@190.205.251.61) |
| 04:21.58 | raden_work | http://www.voltarclamps.com/files/sipconf.txt |
| 04:22.11 | raden_work | http://www.voltarclamps.com/files/sip2.txt <<< last call with that config |
| 04:24.53 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.83) |
| 04:25.42 | raden_work | ok simple question why is it when i call WIFI phone i have bi-directional communication ad when i dial from it i cant hear anything but people can hear me ? |
| 04:28.22 | raden_work | Sending to 192.168.1.120 : 5060 (NAT) |
| 04:28.22 | raden_work | <PROTECTED> |
| 04:28.27 | raden_work | i dont understand this |
| 04:28.38 | raden_work | could something on the netgear be messed up ? |
| 04:29.05 | [TK]D-Fender | localnet=192.168.1.0/255.255.0.0 <- Bad mask |
| 04:30.03 | [TK]D-Fender | under [general] it should be "nat=yes |
| 04:31.50 | raden_work | i dont know how mask got messed up but it fixed nat=yes reload |
| 04:35.56 | raden_work | http://www.voltarclamps.com/files/sip2.txt <<< last call with that config |
| 04:37.58 | raden_work | <-------------> |
| 04:37.58 | raden_work | --- (12 headers 0 lines) --- |
| 04:37.58 | raden_work | Sending to 192.168.1.120 : 5060 (NAT) |
| 04:37.59 | raden_work | <PROTECTED> |
| 04:38.02 | raden_work | i dont get this |
| 04:39.06 | [TK]D-Fender | You showed me another call without configs attached |
| 04:39.17 | [TK]D-Fender | raden_work: You keep focussing on HALF the story. |
| 04:39.34 | [TK]D-Fender | raden_work: Everything in 1 *#$ing pastebin TOGETHER, every time./ |
| 04:39.42 | [TK]D-Fender | randAnd show me you *&$ING FIREWALL |
| 04:39.47 | [TK]D-Fender | raden_work: I asked 5 times already |
| 04:39.58 | raden_work | i did show u the firewall |
| 04:40.11 | raden_work | SERVER:/ # SuSEfirewall2 stop |
| 04:40.11 | raden_work | SuSEfirewall2: batch committing... |
| 04:40.11 | raden_work | SuSEfirewall2: Firewall rules unloaded. |
| 04:40.11 | raden_work | SERVER:/ # |
| 04:40.15 | raden_work | there is the firewall |
| 04:40.19 | raden_work | thats opensuses iptables |
| 04:40.51 | [TK]D-Fender | raden_work: that means jack shit |
| 04:40.58 | [TK]D-Fender | raden_work: iptables --list |
| 04:41.11 | [TK]D-Fender | raden_work: Do I trust some random scheel script? NO |
| 04:41.26 | [TK]D-Fender | shell* |
| 04:41.33 | raden_work | SERVER:/ # iptables --list |
| 04:41.33 | raden_work | Chain INPUT (policy ACCEPT) |
| 04:41.33 | raden_work | target prot opt source destination |
| 04:41.33 | raden_work | Chain FORWARD (policy ACCEPT) |
| 04:41.33 | raden_work | target prot opt source destination |
| 04:41.34 | raden_work | Chain OUTPUT (policy ACCEPT) |
| 04:41.36 | raden_work | target prot opt source destination |
| 04:41.41 | [TK]D-Fender | better |
| 04:41.48 | [TK]D-Fender | and in a pastebin next time |
| 04:42.01 | [TK]D-Fender | only took 6 tries to get this |
| 04:42.24 | raden_work | all u asked for was to show it was stopped basically |
| 04:42.32 | [TK]D-Fender | raden_work: now go restart * and try another call and include your configs |
| 04:42.51 | [TK]D-Fender | raden_work: dump the firewall = show me all the rules. |
| 04:43.10 | raden_work | ok |
| 04:43.16 | [TK]D-Fender | raden_work: You showed me a script message which proves nothing |
| 04:45.58 | KyleK | iptables -t nat --list as well? |
| 04:46.35 | [TK]D-Fender | Why not. |
| 04:47.27 | raden_work | http://www.voltarclamps.com/files/sip.txt |
| 04:47.31 | raden_work | there yeah go |
| 04:48.01 | raden_work | KyleK, that empty as well |
| 04:48.16 | drmessano | IF I am wrong about the Linksys, you should be able to plug the WAN port of the Linksys into the Netgear instead of a LAN port, and your audio will die completely |
| 04:49.03 | raden_work | ill try it just to see what happens |
| 04:53.01 | [TK]D-Fender | NONE of these should eb aon a WAN port |
| 04:53.07 | [TK]D-Fender | be on* |
| 04:53.14 | raden_work | we know |
| 04:53.20 | raden_work | its lan to lan |
| 04:54.23 | [TK]D-Fender | raden_work: Now draw a direct linear line layour of phone1-*-phone2 |
| 04:54.35 | raden_work | huh |
| 04:55.00 | [TK]D-Fender | raden_work: that was pretty clear. Describe every piece of networking from one to the other through * |
| 04:55.16 | [TK]D-Fender | raden_work: in a straight line. This includes IP addresses |
| 04:55.24 | [TK]D-Fender | protocols, media, et |
| 04:55.57 | raden_work | ok |
| 04:57.46 | drmessano | What happened when you tried what I suggested |
| 04:57.48 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 04:57.49 | raden_work | WAN <DSL MODEM > -- Bridged -- <netgear prosafe FV538 192.168.1.1> - #1< asterisk 192.168.1.100> --#2< phone 101> #3<phone 102> #8 < linksys WRT54G 192.168.1.2> Wifi <192.168.1.120> |
| 04:57.57 | raden_work | drmessano, lol |
| 04:58.03 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 04:59.42 | drmessano | ? |
| 04:59.48 | raden_work | what |
| 04:59.55 | drmessano | What happened when you tried what I suggested |
| 05:00.02 | raden_work | absolutley nothing |
| 05:00.36 | raden_work | if i create a route to the net i get same thing 1 way audio |
| 05:00.49 | [TK]D-Fender | raden_work: What firmware is on the WRT? |
| 05:01.04 | raden_work | the newest one from linksys lemee check |
| 05:01.13 | raden_work | <PROTECTED> |
| 05:01.20 | raden_work | newest one i can find |
| 05:01.24 | raden_work | for that model |
| 05:01.34 | drmessano | V6? |
| 05:01.48 | raden_work | WRT54GL V1.1 |
| 05:01.51 | drmessano | ah |
| 05:02.28 | alunca | man, my IP Trunk Registrations just turn to 0 and I cannot make or received call. don't know why. |
| 05:03.57 | [TK]D-Fender | raden_work: So the Linksys is DHCP off, fixed IP, and plugged into a LAN port of the Netgear via one of its own LAN ports? |
| 05:04.04 | raden_work | that is correct |
| 05:04.30 | raden_work | tried that same config with Linksys / ddwrt / tomato |
| 05:04.39 | raden_work | samre results |
| 05:04.52 | drmessano | I think its funny that when you plug in the WAN port, set a route, you get the SAME result as wireless <> LAN |
| 05:05.11 | raden_work | no nothing happens |
| 05:05.20 | raden_work | unless i set a route to the other routers net |
| 05:05.31 | drmessano | Thats what I fuckin said |
| 05:05.33 | raden_work | oh sorry didnt see set a route |
| 05:05.37 | raden_work | dont get soo pissy |
| 05:05.41 | raden_work | yeah we all crabby |
| 05:05.52 | [TK]D-Fender | raden_work: hrm |
| 05:06.49 | [TK]D-Fender | raden_work: Sanity check time |
| 05:06.56 | drmessano | So with a NAT between the Wireless and WAN port, you get 1 way audio.. With the Wireless <> LAN you get 1 way audio.. Hard wired on the very same switch ports, you dont get 1 way audio |
| 05:07.07 | [TK]D-Fender | raden_work: Make an exten that Answer(), then Playback() a file, then Echo() |
| 05:07.18 | [TK]D-Fender | raden_work: test each phone against this independently |
| 05:08.21 | raden_work | no offense but what will that accomplish every phone besides Wifi phone was used today to answer and call and place people on hold etc... |
| 05:08.40 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 05:08.40 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:09.20 | [TK]D-Fender | raden_work: The test should take less time than your argument. That's reason enough |
| 05:09.27 | raden_work | [TK]D-Fender, ill do it just dont see the point |
| 05:10.08 | *** join/#asterisk talntid (n=asd@c-98-247-117-146.hsd1.wa.comcast.net) |
| 05:13.13 | *** join/#asterisk werdan7 (i=w7@freenode/staff/wikimedia.werdan7) |
| 05:14.00 | raden_work | ok now what |
| 05:14.34 | raden_work | i can call that app and hear the file and hear myself |
| 05:14.47 | [TK]D-Fender | raden_work: Fell like describing any of it... |
| 05:14.51 | [TK]D-Fender | raden_work: All phones? |
| 05:14.56 | raden_work | yup |
| 05:14.58 | raden_work | WIFI |
| 05:15.03 | raden_work | and all desk phones |
| 05:15.16 | raden_work | now im really confussed |
| 05:15.27 | [TK]D-Fender | raden_work: ok, with the WIFI go leave a voiccmail. |
| 05:15.36 | raden_work | not setup at moment |
| 05:15.42 | raden_work | i can record a sound file via extension ? |
| 05:15.53 | [TK]D-Fender | raden_work: Actually... here's a thought... jsut ditch teh "answer()" you have before your dial with a NoOp |
| 05:15.54 | raden_work | or that not good enough |
| 05:16.04 | raden_work | in what part ? |
| 05:16.14 | [TK]D-Fender | raden_work: in the extens that dial your phoens. |
| 05:16.29 | [TK]D-Fender | raden_work: You are having * issue an aswer without playing audio then calling another phone |
| 05:16.37 | [TK]D-Fender | raden_work: All before RTp is finalized |
| 05:16.42 | [TK]D-Fender | raden_work: replace that with a NoOp |
| 05:17.04 | raden_work | every single answer in extensions.conf ? |
| 05:17.52 | *** join/#asterisk cobra2599 (n=cobra259@cpe-66-25-60-236.tx.res.rr.com) |
| 05:18.18 | cobra2599 | Anyone know of any modules that have been created for Text to speach through the asterisk API |
| 05:18.58 | raden_work | [TK]D-Fender, wtf i can now speak both ways |
| 05:19.02 | raden_work | on the lan |
| 05:19.26 | [TK]D-Fender | raden_work: As I said, incomplete RTP setup. Test both ways. A> B, B<A |
| 05:19.26 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 05:19.58 | raden_work | ok all lan works but if i dial out cant hear anyone |
| 05:20.06 | tengulre | hi,all |
| 05:20.21 | [TK]D-Fender | raden_work: we're only dealing with LAN+WIFI right now |
| 05:20.26 | [TK]D-Fender | raden_work: One problem at a time |
| 05:20.32 | raden_work | if i dial inbound with my cellphone i have bi-directional |
| 05:20.33 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 05:20.38 | raden_work | if i dial out with wifi i cant hear |
| 05:20.47 | [TK]D-Fender | raden_work: CONFIRM |
| 05:21.03 | raden_work | CONFIRM > |
| 05:21.18 | [TK]D-Fender | raden_work: Is that? |
| 05:21.27 | raden_work | is what > |
| 05:21.33 | [TK]D-Fender | ........... |
| 05:21.44 | [TK]D-Fender | raden_work: Why am I hearing about your CELL PHONE |
| 05:21.46 | raden_work | on the lan all works |
| 05:21.57 | raden_work | if i dial out i cannot hear |
| 05:22.01 | [TK]D-Fender | radeqI asked you to confirm bi-directional audio on ALL LOCAL CALLS |
| 05:22.05 | raden_work | dialing in from cell have bi-dreictional |
| 05:22.12 | [TK]D-Fender | raeI did not say "try other outside shit" |
| 05:22.17 | raden_work | all lan calls yes bi-directional |
| 05:22.21 | [TK]D-Fender | FUCK THE CELL PHONE! |
| 05:22.26 | raden_work | wow fine |
| 05:22.30 | raden_work | not enough info |
| 05:22.33 | [TK]D-Fender | Jesus H Christ we are jsut dealing with your SIP PHONES! |
| 05:22.33 | raden_work | to much jesus |
| 05:22.35 | tengulre | how to use transfer under agent mode ? |
| 05:22.39 | raden_work | just yell at me some more |
| 05:22.57 | [TK]D-Fender | raden_work: I'm trying to get a straight story from you and you are jumpiong on to other topics |
| 05:23.15 | [TK]D-Fender | You are not confirming where you have gotten yourself clearly before running off on tangents. |
| 05:23.18 | raden_work | I said for time #3 all internal phones on lan have bi-directional communication |
| 05:23.28 | [TK]D-Fender | No wonder people are getting lsot rying to help you. |
| 05:23.33 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-esyunbassoddgvud) |
| 05:23.37 | raden_work | cause people cant freaking read apparently |
| 05:23.47 | [TK]D-Fender | raden_work: and Bi directional communication may depends on WHO IS CALLING WHO |
| 05:23.47 | raden_work | <raden_work> ok all lan works but if i dial out cant hear anyone |
| 05:23.58 | raden_work | ALL |
| 05:23.59 | [TK]D-Fender | raden_work: The directions you tested were not explicit |
| 05:24.08 | raden_work | as in every freaking phone calling every freaking phone works |
| 05:24.12 | [TK]D-Fender | raden_work: Fine, one thing down. |
| 05:24.17 | [TK]D-Fender | \o/ |
| 05:24.20 | raden_work | ok |
| 05:24.39 | cobra2599 | i ran into that problem before im sure it is a trunk setting, i believe nat=yes fixed it for me on the trunk settings |
| 05:24.42 | [TK]D-Fender | raden_work: pastebin a complete call attempt to the outside |
| 05:24.51 | drmessano | lol |
| 05:25.10 | drmessano | Youre out of your league, donnie |
| 05:26.01 | raden_work | http://pastebin.com/dc2f14ea |
| 05:26.19 | raden_work | cobra2599, its only 1 phone |
| 05:27.04 | [TK]D-Fender | raden_work: Reliably Transmitting (NAT) to 204.11.192.36:5060: |
| 05:27.17 | [TK]D-Fender | raden_work: Your callcentric peer should be nat=no like the guide told you |
| 05:28.03 | cobra2599 | ah so i had it backwards its been a while lol |
| 05:28.12 | cobra2599 | at least i was close |
| 05:28.53 | raden_work | [TK]D-Fender, did that still not workign |
| 05:29.20 | raden_work | cobra2599, you were more than likley correct |
| 05:29.33 | [TK]D-Fender | raden_work: And every time I hear that I should be seeing a pastebin to match with new configs |
| 05:29.41 | [TK]D-Fender | raden_work: I should not have to ask for this every time |
| 05:29.57 | raden_work | i change one lil thing and you need to see it ? |
| 05:30.14 | [TK]D-Fender | ITSP entries should almost never be "nat=yes" |
| 05:30.21 | [TK]D-Fender | raden_work: Correct. |
| 05:30.57 | [TK]D-Fender | raden_work: If people did things right all the time like they swear they do, it would typically work <- |
| 05:30.57 | raden_work | http://pastebin.com/d246f3323 |
| 05:31.04 | cobra2599 | Anyone know of any modules that have been created for Text to speach through the asterisk API |
| 05:31.25 | raden_work | cobra2599, contact digium |
| 05:32.14 | raden_work | [TK]D-Fender, why is it that all the other phones here work but not 1 lil wifi phone ? i dont see how all this could be wrong ? |
| 05:33.05 | [TK]D-Fender | raden_work: that is not a compelte call, and I don't see your configs in there. |
| 05:33.23 | [TK]D-Fender | raden_work: Why is this so difficult to follow? |
| 05:34.10 | raden_work | 1100 lines and its not a complete call |
| 05:34.42 | [TK]D-Fender | raden_work: the pastebin STARTS with a 183 PROGRESS.. that means the PREVIOUS messages were the start of the cal |
| 05:34.44 | [TK]D-Fender | call |
| 05:34.55 | [TK]D-Fender | raden_work: And still no configs |
| 05:35.15 | raden_work | http://pastebin.com/d70d34b03 |
| 05:35.53 | raden_work | omg you need my config to make sure i inserted nat=no under [callcentric] ??? |
| 05:37.13 | [TK]D-Fender | raden_work: And to see if if you screewed other stuff up |
| 05:37.29 | raden_work | omg this is ridiculous cause im such a F up |
| 05:38.20 | raden_work | http://pastebin.com/d31a8a513 |
| 05:38.26 | raden_work | there tell me everything is screwed up |
| 05:38.26 | [TK]D-Fender | raden_work: You are doing everything in your power to NOT look. Thats what's fucked up. You put up a fight for the simplest stuff when poelple have wasted substantial amounts of time trying to keep you from running in circles. |
| 05:38.44 | raden_work | whatever |
| 05:40.29 | [TK]D-Fender | directrtpsetup=yes <- ditech |
| 05:40.31 | [TK]D-Fender | ditch |
| 05:40.59 | [TK]D-Fender | Most of the rest looks largely right |
| 05:41.26 | [TK]D-Fender | instead of exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@callcentric) |
| 05:41.36 | [TK]D-Fender | try exten => _1NXXNXXXXXX,1,Dial(SIP/callcentric/${EXTEN}) |
| 05:42.02 | [TK]D-Fender | And I might consider setting the callerID before calling out <_ |
| 05:42.06 | raden_work | that would make more sense |
| 05:42.17 | raden_work | howso ? |
| 05:42.40 | [TK]D-Fender | Wait... fromuser should override that... |
| 05:42.49 | [TK]D-Fender | radejust try this last thing |
| 05:42.58 | [TK]D-Fender | raden_work: and make sure to do verbsoe 10 |
| 05:43.04 | raden_work | the last thing |
| 05:43.13 | [TK]D-Fender | [01:41]<[TK]D-Fender>try exten => _1NXXNXXXXXX,1,Dial(SIP/callcentric/${EXTEN}) |
| 05:43.17 | [TK]D-Fender | ^^^ |
| 05:43.42 | *** join/#asterisk fiddur (n=fiddur@192.121.104.122) |
| 05:44.12 | drmessano | 120, nat=yes |
| 05:44.32 | raden_work | http://pastebin.com/d297061ef |
| 05:44.43 | [TK]D-Fender | UGH |
| 05:44.50 | [TK]D-Fender | indeed. |
| 05:45.03 | [TK]D-Fender | raden_work: WTF is [120] doing with "nat=yes"? |
| 05:45.16 | drmessano | Talk about not listening |
| 05:46.15 | raden_work | its set to no now |
| 05:46.22 | [TK]D-Fender | raden_work: Correct that and test my revised format Dial |
| 05:47.05 | *** join/#asterisk maour (n=quassel@unaffiliated/maour) |
| 05:47.55 | raden_work | http://pastebin.com/d297061ef starts at line 481 |
| 05:49.03 | raden_work | wtf |
| 05:49.05 | raden_work | hold on |
| 05:50.11 | raden_work | http://pastebin.com/d4a924a3e |
| 05:50.12 | raden_work | there we go |
| 05:51.20 | [TK]D-Fender | raden_work: What have you forwarded on your Negear to *? |
| 05:51.30 | raden_work | yes |
| 05:51.34 | raden_work | 5060 - 20000 |
| 05:51.37 | raden_work | udp |
| 05:51.57 | [TK]D-Fender | raden_work: Disable any SIP-aware features, etc |
| 05:52.18 | [TK]D-Fender | raden_work: and A word of warning, netgear routers have been known NAT offenders before |
| 05:52.23 | raden_work | to my knowledge there are non on that router i have looked |
| 05:52.35 | raden_work | [TK]D-Fender, your the millionth person to tell me that |
| 05:52.43 | raden_work | u think its the router ? |
| 05:53.18 | raden_work | lemee see if there a firmware update |
| 05:53.53 | [TK]D-Fender | raden_work: There is a very real chance it is |
| 05:54.07 | [TK]D-Fender | raden_work: D-Link & Netgear = trouble |
| 05:54.16 | [TK]D-Fender | Linksys = generally none |
| 05:55.46 | raden_work | omg there have been a ton of firmware updates |
| 05:57.04 | [TK]D-Fender | raden_work: Go test them. |
| 05:57.07 | [TK]D-Fender | Beg time.... |
| 05:57.11 | [TK]D-Fender | back tomorrow |
| 05:57.16 | [TK]D-Fender | Bed* |
| 05:57.19 | raden_work | same here thanks |
| 05:57.19 | [TK]D-Fender | askldhkjajhakljdfgfd |
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| 06:14.26 | raden_work | i updated my firmware and reset everything now im getting [Aug 18 01:13:03] NOTICE[8949]: chan_sip.c:9489 sip_reg_timeout: -- Registration for '17772445766@callcentric.com' timed out, trying again (Attempt #26) |
| 06:14.27 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:16.01 | sergee | Can somebody explain me the purpose of "Caller ID presentation" and the meaning of it's values? |
| 06:16.01 | sergee | Please |
| 06:20.31 | florz | It isn't values, it is a protocol field. |
| 06:21.26 | *** join/#asterisk davidandgoliath (n=David@bowness-out.eng.telusmobility.com) |
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| 06:23.27 | sergee | florz: field contain values, right? |
| 06:24.37 | raden_work | [Aug 18 01:23:14] NOTICE[8949]: chan_sip.c:9489 sip_reg_timeout: -- Registration for '17772445766@callcentric.com' timed out, trying again (Attempt #27) |
| 06:24.50 | raden_work | anyone have any idea why this would be happeneing ? |
| 06:24.58 | raden_work | all i did was upgrade my router firmware |
| 06:25.39 | florz | sergee: well, yeah, usually. But bottles contain water, still they aren't water. |
| 06:25.59 | sergee | florz: "'Presentation Allowed, Not Screened" - what's the meaning of this? what's the difference between 'presentation allowed' and 'presentation prohibited' ? how isdn switches treat this info? |
| 06:26.23 | sergee | florz: well, the question wasn't about field, it was about values |
| 06:27.09 | florz | sergee: that depends on their configuration - but usually, the switch facing the customer drops the caller id if it's flagged as "presentation prohibited" |
| 06:28.45 | sergee | florz: and what's the meaning of 'not screened'? what's the difference between 'not screened' and 'passed screen' ? |
| 06:29.14 | florz | sergee: well, the indicates whether the customer-facing switch checked the callerid for correctnes |
| 06:29.28 | florz | (where the call entered the telco's network) |
| 06:29.40 | sergee | florz: thank you very much@ |
| 06:29.45 | sergee | ! |
| 06:31.03 | sergee | florz: trying to hide callerid in proper way between my asterisk and cisco, but looks like cisco ignoring privacy in "remote-Party-ID" header, i thought i misunderstood privacy meaning.. |
| 06:35.10 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
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| 06:42.22 | raden_work | can someone tell me why my provider would be unreachable ? |
| 06:42.28 | raden_work | i can traceroute just fine |
| 06:43.48 | kaldemar | traceroute doesn't mean that any other protocol would work |
| 06:43.59 | kaldemar | it's just ICMP |
| 06:44.20 | florz | no, it's not |
| 06:45.28 | raden_work | how do i make asterisk like flush its host its trying to goto 204.11.192.38 |
| 06:45.37 | kaldemar | ok, it can be whatever, but it's not VoIP. |
| 06:45.37 | raden_work | when 204.11.192.36 is reachable |
| 06:48.16 | florz | but it's usually udp, which is damn close to voip, kindof |
| 06:48.44 | raden_work | well why is my host unreachable ? |
| 06:49.11 | florz | however, it's icmp for the reply, so an icmp problem can interfere with traceroute |
| 06:50.25 | kaldemar | close and close. still different port, different software stack... |
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| 06:53.33 | raden_work | [Aug 18 01:52:11] NOTICE[9527]: chan_sip.c:9489 sip_reg_timeout: -- Registration for '17772445766@callcentric.com' timed out, trying again (Attempt #1) |
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| 07:31.47 | khussein78 | how can i configure SIP Trunk without username and secret |
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| 08:13.32 | thisismyname | why would u do that? |
| 08:13.55 | thisismyname | @khussein78 |
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| 09:00.41 | rjek | Oh, wrong channel, I want #asterisknow. Sorry for the noise. |
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| 09:06.51 | stix | When using the asterisk command "Monitor", will this only record/monitor one channel? I mean if I monitor a call, will I only hear sound from one party when playing the file? |
| 09:07.58 | [netman] | both parties |
| 09:08.05 | stix | great |
| 09:12.10 | *** join/#asterisk unasi7 (n=unasi7@80-218-32-110.dclient.hispeed.ch) |
| 09:12.37 | michael-i | I'm currently porting my Asterisk based project over to run on a Blackfin based hardware. Compilation goes fine, running asterisk -c on the device loads all of the modules but bails out after res_monitor.so with: |
| 09:12.44 | michael-i | asterisk: can't resolve symbol '_dialed_interface_info' |
| 09:14.48 | michael-i | I've know what's happening there but not why ( of course ). Reference to that symbol is only in app_queue and app_dial |
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| 09:19.33 | kaldemar | stix: monitor generates separate files for the channel's input and ouput. use option m or MixMonitor to get them to one file. |
| 09:19.58 | stix | kaldemar, thanks - important info |
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| 09:32.49 | stix | kaldemar, I have this "exten => s,n,Monitor(wav,${FILNAVN},m)" but still I get ...in.wav and out.wav files ? |
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| 09:34.44 | tzafrir_laptop | michael-i, don't you use statically-linked asterisk? |
| 09:35.23 | michael-i | tzafrir_laptop: nope...just found another unresolved symbol |
| 09:35.25 | stix | kaldemar, sorry, just me being impatient |
| 09:35.27 | michael-i | let the fun begin |
| 09:35.39 | tzafrir_laptop | If so, res_monitor should be linked in . There should be no need to load an external res_monitor.so |
| 09:37.08 | michael-i | it's not actually res_monitor that's failing. I tried moving app_dial.so out of the modules dir which resulted in the next module failing on another symbol (func_math.so ___ast_module_user_remove) |
| 09:44.45 | inckie | im prette new to asterisk, and i trying to get a snomsoft phone to reg on my asterisk |
| 09:45.00 | inckie | but just get snomsoft1@shptest.hgradio.dk: Registration failed |
| 09:45.16 | inckie | this is my configuration in sip.conf |
| 09:45.17 | inckie | http://pastebin.ca/1533596 |
| 09:54.52 | *** join/#asterisk Geert (i=geert@irssi/staff/geert) |
| 09:55.07 | Geert | Reason: Q.850;cause=3;text="NO_ROUTE_DESTINATION" => Can I play a custom message when receiving this? (SIP) |
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| 10:00.43 | Tar-Get | hi, I installed asterisk 1.6.1.4 with freepbx |
| 10:01.06 | Tar-Get | when I reload config changes asterisk crashes |
| 10:01.11 | Tar-Get | http://pastebin.com/m7c388ab |
| 10:01.23 | Tar-Get | that is the core dump |
| 10:01.46 | Tar-Get | can someone help me solve the problem? |
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| 10:08.12 | Vec | Hello, I have an Avaya 9620, running the SIP firmware R2.4, it works great with Asterisk except one small hangup, when the SIP registration timeout occurs, and it attempts to re-register, it displays a login incorrect msg and then asks the user for the user and pass, even if its entered correctly it says login failed. |
| 10:09.09 | *** part/#asterisk Tar-Get (n=Tar-Get@83.101.83.38) |
| 10:09.30 | kaldemar | Geert: sure, use function SIP_HEADER to inspect the header and then proceed accordingly in your dialplan. |
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| 10:12.49 | Geert | kaldemar: tnx; will look into it |
| 10:17.20 | semaries | Hello, does anyone of you know the Grandstream GXW-400X box? |
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| 10:21.02 | alunca | anyone here own a WRT54GP2A-AT ? |
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| 10:41.52 | Geert | kaldemar: You can't use that on a reply from the server. (SIP provider) |
| 10:42.15 | Geert | SIP_HEADER() is the SIP info of the initiating client/server |
| 10:45.57 | kaldemar | ${HANGUPCAUSE} might do the job then, if it gets set right |
| 10:46.21 | Geert | Nope; I already tried that |
| 10:46.56 | Geert | $HANGUPCAUSE returns 1 |
| 10:54.13 | Geert | kaldemar: I figured it out; tnx :) |
| 10:54.23 | Geert | I found a correct hangupcause table |
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| 11:15.43 | *** join/#asterisk Bach (n=bach@exchange.fuzion.dk) |
| 11:15.58 | Bach | Hi... Asterisk Newbee here |
| 11:17.19 | Bach | Looking for a setup, so i can use asterisk as a Gateway server for several other VOIP servers. Meaning that this should only connect 1 outbound/inbound trunk, and then have more asterisk trunks connected, so it can distribute calls to them. So this server should not have any "users" or extentions, is that possible |
| 11:22.22 | kaldemar | Bach: yes |
| 11:22.54 | Bach | been looking everywhere for documentation, can you point me the right direction ? |
| 11:23.29 | kaldemar | ~book |
| 11:23.30 | 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 |
| 11:23.37 | kaldemar | ~wikis |
| 11:23.38 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 11:23.54 | kaldemar | start with those, the book first |
| 11:24.56 | Bach | okay, what is it called in the correct terms Gateway server or something like that ? |
| 11:29.58 | *** join/#asterisk hershel (n=hershel@213.8.21.65) |
| 11:31.14 | kaldemar | for example gateway. depends on the usage. |
| 11:31.27 | hershel | Newbie question, if I may. I just installed asterisk 1.4 binary via apt-get on a Debian stable box, but in /var/lib/asterisk there is no agi-bin directory. Seems from all the tutorials that there should be one there with sample agi files |
| 11:31.43 | *** join/#asterisk PuroOsso (n=PuroOsso@189.74.141.10) |
| 11:38.11 | leifmadsen | hershel: just search for agi -- I think the debian binaries puts things in non-standard places |
| 11:38.31 | Bach | kalemar: At this time, i have a Swyx setup, connected to a Cisco media gateway, i want to place an Asterisk in between theese two, so the Cisco MGW talks with Asterisk, and the Asterisk routes the calls for the SWYX setup. At a later time, I'll connect more numbers and needs them diverted to another asterisk server. So i need the Asterisk server between CMGW and SWYX |
| 11:42.43 | *** part/#asterisk robot12 (n=robot12@78.138.154.66) |
| 11:45.27 | hershel | leifmadsen, thanks but " find / -name agi " brings no results. :( searching for *.agi brings up one file in /usr/share/doc/asterisk-config/examples But I don't think it's supposed to execute from there. :) |
| 11:46.03 | leifmadsen | well, I don't really "believe" in using distributed binaries as they are usually behind the releases |
| 11:46.09 | leifmadsen | and it's not that hard to compile really |
| 11:46.30 | hershel | yes i figured that's the answer i would get. :( lol |
| 11:46.59 | hershel | i dont' need advanced features--just to route incoming calls to AGI. but i suppose i should compile, eh? |
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| 11:54.48 | skrusty | hershel: you'll often find even when you dont want the cutting edge features, the binaries lag behind in small, yet important updates |
| 11:55.05 | skrusty | for example, i installed from debian apt, whihc is on 1.2.14 i think |
| 11:55.09 | *** join/#asterisk garymc (n=gar@host81-134-0-102.in-addr.btopenworld.com) |
| 11:55.16 | skrusty | but that didn't include a fix for flock on smbfs |
| 11:55.26 | skrusty | so i had to go from src |
| 11:55.32 | garymc | Hi, can i put asterisk on a Ubuntu 9.04 LTSP Server? |
| 11:56.00 | garymc | I just want to link this Asterisk server to my LTSP clients on my other server and give them all a voip phone etc |
| 11:56.05 | garymc | can i do this? |
| 11:56.10 | garymc | what package should i install |
| 11:58.15 | *** join/#asterisk aurax (n=aurax@bzq-179-76-199.static.bezeqint.net) |
| 11:58.41 | hershel | debian apt has 1.4.21.2 actually (now at least) but i hear. truth is it should have had the agi-bin directory. could't i just make that directory? in theory at least? :) |
| 11:59.58 | skrusty | yeah of course |
| 12:00.07 | skrusty | i use fast agi now |
| 12:00.25 | skrusty | means i can keep all my AGIs on a dedicated box, away from * |
| 12:04.14 | hershel | fastagi? Hmmm I see online what the difference is. Doesnt seem any more complex. Let me explain what I'm doing. I am a php programmer and I am making this box so that a store (my client) can have a phone number which people can call and enter their order number and then hear the status of that order played back. That's it. fairly simple. |
| 12:04.40 | skrusty | cool, yeah pretty simple |
| 12:04.42 | hershel | the box won't be used for anything else. and I myself am a complete newbie to Asterisk. So I suppose the easiest route makes sense. :) |
| 12:05.34 | skrusty | well, if it's just that, and all you intend to do with * is that, i wouldn't suggest complicating it with multiple boxes (i.e. * server and a FastAGI server) unless you can reuse both for other projects |
| 12:06.03 | skrusty | if i were you, id have my pho agi script on the same box |
| 12:06.13 | skrusty | php even |
| 12:06.32 | hershel | ok. good. i would learn to program in pho but who has time? ;) |
| 12:06.40 | skrusty | hehe :) |
| 12:06.53 | skrusty | i do my AGI's in .NET |
| 12:07.04 | skrusty | and although there is mono (whihc is great) i use linq |
| 12:07.32 | hershel | i plugged a line into the box (brand new TDM410P with FXO) and called it but it doesn't answer. :( |
| 12:07.43 | hershel | .NET? ick. |
| 12:07.56 | skrusty | dont knock the dot ;) |
| 12:08.15 | hershel | seems from extensions.conf that by default it should answer and same congrats. no ? |
| 12:08.17 | skrusty | does it show in your zap channels? |
| 12:08.20 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 12:08.33 | skrusty | and is the default context for that zap channel 'default'? |
| 12:08.39 | *** join/#asterisk lowtek (n=anonymou@mail.heavylogic.com) |
| 12:11.12 | hershel | does what show in my zap channels? how do i check this exactly? in zapata.conf ? |
| 12:11.35 | tzafrir_laptop | skrusty, use FastAGI and send it to some remote port |
| 12:11.46 | tzafrir_laptop | see core show application FastAgi |
| 12:11.49 | skrusty | tzafrir_laptop: ? sorry? |
| 12:12.14 | skrusty | i do already, im not stuck, i was explaining my setup and needs to hershel :) |
| 12:12.16 | tzafrir_laptop | ah, ok. I thought you asked how to do that |
| 12:12.21 | skrusty | hehe :) |
| 12:12.37 | skrusty | hershel: what version are you running? |
| 12:12.42 | skrusty | of * |
| 12:12.54 | hershel | <PROTECTED> |
| 12:13.05 | tzafrir_laptop | hershel, it's in /usr/share/asterisk/agi-bin by default, there |
| 12:13.20 | tzafrir_laptop | The datadir in Debian is /usr/share/asterisk |
| 12:13.59 | hershel | tzafrir, Shalom, it didn't make the agi-bin dir. there |
| 12:14.01 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 12:14.34 | tzafrir_laptop | can you make one? |
| 12:14.39 | *** join/#asterisk coppice (n=chatzill@52.204.17.210.dyn.pacific.net.hk) |
| 12:15.02 | skrusty | you can make it anywhere and set the location in asterisk.conf i think |
| 12:15.04 | skrusty | iirc :) |
| 12:15.08 | hershel | Yes, but first issue (I think) is to get this box to answer the phone. :) |
| 12:15.44 | skrusty | hershel: not used any zap devices for a few years |
| 12:15.48 | hershel | i have TDM410P with FXO with a VOIP line plugged in. i just tested the VIOP line (from sipura spa 2K) and it works with a regular phone |
| 12:17.37 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
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| 12:19.41 | *** join/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 12:24.46 | leifmadsen | VoIP comes on specific lines now? |
| 12:26.08 | hershel | Huh? What I mean is that I have a VOIP box (a Sipura SPA 2000) Out of that comes a phone socket which has a dialtone. I plug into there a regular PSTN phone and call the number and it works. |
| 12:26.26 | hershel | but if i plug in instead a wire from that socket into my new FXO and call then it doesn't answer. |
| 12:27.21 | tzafrir_laptop | Speaking of Debian: the latest Debian-based live CD / USB: http://updates.xorcom.com/iso/ |
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| 12:28.59 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 12:31.55 | Skeeter- | Hi |
| 12:31.57 | Skeeter- | anyone got asterisk working along with freepbx on ubuntu 9.04 or debian?? |
| 12:32.11 | garymc | im trying too get something working |
| 12:32.25 | garymc | im installing ubuntu now and i want Astlinux to work on it |
| 12:32.36 | [TK]D-Fender | Skeeter-: I'm sure a lot of people do |
| 12:32.46 | garymc | not sure what version of asterisk to download or how i would get it onto my ubuntu server? |
| 12:32.48 | *** join/#asterisk LiNeTuX (n=LiNeTuX@fl-209-26-240-157.sta.embarqhsd.net) |
| 12:32.52 | garymc | what should i apt-get |
| 12:33.00 | lowtek | $@#(@&*($(&@$ -> be sure to Clear your MySQL results properly if using MySQL() or your sh*t will eventually crash (1.4.26) |
| 12:33.11 | garymc | should i just type "sudo apt-get asterisk" |
| 12:33.11 | lowtek | type-o's suck |
| 12:33.12 | Skeeter- | i got everything installed proprely i think, both serviices are running but i cant access the freepbx panel |
| 12:33.30 | [TK]D-Fender | Skeeter-: FreePBX is NOT supported here |
| 12:33.32 | [TK]D-Fender | ~freepbx |
| 12:33.33 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 12:33.52 | Skeeter- | garymc: i suggest you to install everything manually |
| 12:33.57 | garymc | how? |
| 12:34.12 | garymc | you install manually? |
| 12:34.15 | [TK]D-Fender | garymc: Download the source tarball and compile like the rest of us |
| 12:34.43 | lowtek | garymc: compiling from source is *really* easy |
| 12:35.00 | garymc | you wanna talk me through in a bit once ubuntu is instaslled? |
| 12:35.12 | Skeeter- | well |
| 12:35.22 | garymc | :) |
| 12:35.22 | lowtek | garymc: Just google "install asterisk ubuntu" and there's 3 or 5 really good walkthroughs ... |
| 12:35.29 | garymc | ok |
| 12:35.41 | Skeeter- | simply type: asterisk freepbx 2.5 ubuntu |
| 12:35.44 | Skeeter- | in google |
| 12:35.55 | fiddur | cd asterisk; less README; |
| 12:36.07 | Skeeter- | get the tut for server 9.04, simply download the headers for the desktop version instead |
| 12:40.25 | Naikrovek | garymc: http://www.voip-info.org/wiki/view/Asterisk+Linux+Ubuntu?utm_source=voip-info&utm_medium=module&utm_campaign=recentchanges |
| 12:41.27 | garymc | i seen this one Naikrovek but it doesnt have everything with it |
| 12:41.33 | garymc | it says that at the start |
| 12:41.39 | garymc | surely i want everything? |
| 12:42.51 | Naikrovek | do you want a system to play with or is this something you're going to use in production somewhere |
| 12:43.04 | hershel | I ran genzaptelconf on my debian box and it finds ztdummy but not my 4 channel card that has one FXO in it. I though it was supposed to detect the card, heh heh. |
| 12:43.52 | garymc | im going to use it in production soon ish |
| 12:43.56 | garymc | well as soon as possible |
| 12:44.08 | garymc | but prob use somethign to play with first. I need to record all calls too |
| 12:44.23 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 12:46.04 | michael-i | I'm still working on my blackfin cross-compile. All loadable modules are failing with unresolved symbol errors. I just noticed that the error is unresolved "___ast_module_user_add" (3 underscores) and strings+grep on the asterisk binary shows "__ast_module_user_add" (2 underscores) |
| 12:47.00 | michael-i | just throwing that out there....I'm still clueless here |
| 12:51.27 | *** part/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 12:51.40 | hershel | i rebooted and ran genzaptelconf and it found my TDM 410P and it says fxsks=1 for my FXO mod. but i think that may be right. Is that correct or should it be fxoks ? |
| 12:51.54 | [TK]D-Fender | hershel: It is correct |
| 12:52.25 | hershel | Hey, this is pretty cool. :) Thanks. (who ever said asterisk isn't fuN? ;) ) |
| 12:53.16 | hershel | Oh, now I found it: FXO ports use FXS signaling. The fxsks indicates that it is a FXO port with kewlstart signalling. OK< I will note this. |
| 12:54.21 | [TK]D-Fender | hershel: FXO card is meant for you to talk to the OFFICE. You are therefore the STATION. That's the way to think of this |
| 12:54.25 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 12:55.20 | hershel | hmmm. ok. i am going to plug in PSTN line into my FXO and let ppl call (and then process their calls with an AGI script). i guess they're the office, then eh? |
| 12:57.00 | *** join/#asterisk LiNeTuX (n=LiNeTuX@fl-209-26-240-157.sta.embarqhsd.net) |
| 12:58.19 | tzafrir_laptop | michael-i, any chance that this is a custom module that was written for 1.4 and you build 1.6.x ? |
| 12:58.26 | [TK]D-Fender | hershel: The telco is the "Office" actually |
| 12:58.27 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) |
| 12:59.09 | hershel | oh. makes sense. now i see that my box has no zapata-auto.conf file. nowhere. Seems genzapconf didn't make one. :( |
| 12:59.53 | tzafrir_laptop | zapata-channels.conf ? |
| 13:00.35 | hershel | yeah, i have one of those. |
| 13:02.28 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 13:02.45 | hershel | Thank you to all who helped me. I must go now. I may be back later. :) |
| 13:04.44 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 13:05.50 | michael-i | tzafrir_laptop: nope. these are the standard 1.4.26 modules |
| 13:07.23 | fiddur | If I use linear queue strategy with realtime queue members; will it ring the queue members in order of uniqueid then, or isn't linear supported for realtime? |
| 13:09.53 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 13:10.11 | fiddur | ...or qill I have to reimplement the linear strategy using penalty.... |
| 13:11.30 | [TK]D-Fender | fiddur: There is no "uniqueid" for members |
| 13:12.15 | lowtek | "How to bring asterisk to it's knees with a simple typ-o", by LowTek |
| 13:12.25 | lowtek | ARHG&@(&$@$ s/typ-o/type-o |
| 13:12.28 | lowtek | mf |
| 13:12.28 | fiddur | [TK]D-Fender: In the realtime table there is a uniqueid... it has to be, otherwise app_queue can't change pause-status... |
| 13:12.34 | ruben23 | hi--> anyone get bad audio quality for codec alaw & ulaw with xlite softphones on linux installed...? |
| 13:13.08 | [TK]D-Fender | fiddur: Interesting... |
| 13:14.05 | fiddur | [TK]D-Fender: And there is one in the http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue and in the contrib/scripts/realtime_pgsql.sql |
| 13:14.26 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:14.41 | fiddur | [TK]D-Fender: and in app_queue.c: char rt_uniqueid[80]; /*!< Unique id of realtime member entry */ |
| 13:16.14 | [TK]D-Fender | fiddur: While it may use that so as to be sure to keep state on the right member, I don't it's using it for sequencing the call-outs |
| 13:18.36 | fiddur | [TK]D-Fender: I don't think so either... but I was hoping there was something in the realtime layer that ordered it somehow... but I guess not |
| 13:20.31 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:20.58 | *** join/#asterisk glam (n=glam@120.86.214.143) |
| 13:22.40 | *** join/#asterisk GeminiDomino (n=C@fl-71-0-246-138.sta.embarqhsd.net) |
| 13:23.34 | skrusty | ruben23: i use x-lite as a test client, works fine with alaw and ulaw |
| 13:24.04 | skrusty | ruben23: how is the call quality degraded, jitter broken audio? |
| 13:24.47 | ruben23 | skrusty:yes it broken voice and choppy....but with windows its fine..are there any advance settings do i need for xlite or for the linux distro.. |
| 13:25.16 | skrusty | ruben23: ah, only used it on windows |
| 13:25.18 | skrusty | soz :/ |
| 13:27.28 | *** join/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 13:27.36 | *** join/#asterisk Lyma (n=Lyma@unaffiliated/lyma) |
| 13:28.11 | ruben23 | skrusty:problem now---linux are rapidly deployed on my network changing all the windows client |
| 13:29.27 | [TK]D-Fender | Whats the point of running a closed softphone on Linux? Use Ekiga instead |
| 13:31.02 | ruben23 | ok i will test Ekiga--hope it will solve the voice issues |
| 13:32.09 | *** join/#asterisk shido6 (n=shido6@74.132.202.71) |
| 13:32.32 | lowtek | On Ekiga, how is it overall? Would you use it say in a small call center? |
| 13:32.37 | *** join/#asterisk jmacz (n=mcorb@190.144.75.22) |
| 13:33.08 | [TK]D-Fender | lowtek: What makes any softphone that much fidderent than another? |
| 13:33.28 | [TK]D-Fender | lowtek: I would never subject my coworkers to any softphone |
| 13:34.06 | Pan3D | lowtek: play with it and determine that for yourself. You know your customers. |
| 13:34.19 | GeminiDomino | Hey, here's a weird one... Is there any way to lower the noise detection so that WaitForSilence doesn't loop until timeout regardless of what happens? I just used my cell phone to monitor the script, and the silence timer keeps resetting, even if the phone is muted |
| 13:34.21 | Pan3D | Ekiga is relatively easy to figure out for geeks |
| 13:34.51 | lowtek | TK: I've never used Ekiga, but we see Bria used with a lot of success and X-Lite given up on easily. I don't use one myself, I use a 330 with a headset jacked in the side -- best solution imho. |
| 13:35.46 | Pan3D | lowtek: what OSes are being used in the call center? |
| 13:36.30 | leifmadsen | anyone had an issue with calls between DAHDI and chan_sip dropping after 30 seconds when you place a call on hold? |
| 13:36.34 | lowtek | Pan3D: I was looking for a generalization on the state of Ekiga ... The call center customers we have use Bria or just Polycom's with headsets. |
| 13:36.35 | leifmadsen | (using Asterisk 1.6.2) |
| 13:36.39 | [TK]D-Fender | lowtek: I run IP 600's w/ Plantronics M22 amps & H261n binaural headsets |
| 13:36.55 | leifmadsen | currently going through the .txt files, and will check out the bug tracker shortly -- just thought I'd check if anyone had run into that before. |
| 13:36.56 | [TK]D-Fender | leifmadsen: Which side drops? |
| 13:36.59 | lowtek | TK: $$! :) |
| 13:37.12 | [TK]D-Fender | leifmadsen: I have head of sip devices that freak about not getting RTP... |
| 13:37.16 | Pan3D | lowtek: have you run into problems with the Polycoms? |
| 13:37.29 | lowtek | Pan3D: No way, best phone around!!! |
| 13:37.37 | leifmadsen | [TK]D-Fender: it's hard to tell in the console debug... I *think* it's the SIP side. And ya, this is usually an RTP issue, although it seems odd with it being a call between DAHDI and SIP |
| 13:37.38 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 13:37.39 | [TK]D-Fender | lowtek: Yes, and worth every penny. Clarity & comfort for people who spend all day on the phone. |
| 13:37.46 | Pan3D | k, I was going to say, that would be suprising |
| 13:38.00 | [TK]D-Fender | lowAlso no distracting client to pop up or have to manipulate |
| 13:38.09 | Naikrovek | polycom phones are better than cisco phones, i'd say |
| 13:39.06 | Pan3D | softphones can take up precious screen realestate if you're trying to get stuff done (e.g. handle calls). I'd stick with the polycoms if they workd. |
| 13:39.09 | [TK]D-Fender | leifmadsen: Other freakish idea is if silence detection & call progress is an issue if its a POTS call |
| 13:39.10 | lowtek | Naikrovek: I think most in channel would agree where asterisk is concerned. |
| 13:39.21 | leifmadsen | [TK]D-Fender: PRI |
| 13:39.41 | [TK]D-Fender | leifmadsen: Well that seriously makes the SIP side suspect |
| 13:40.01 | leifmadsen | [TK]D-Fender: that's what I figured, but I'm trying to think why it thinks it should hangup |
| 13:40.02 | leifmadsen | http://pastebin.ca/1533817 |
| 13:40.03 | Naikrovek | lowtek: yeah i've not used any cisco solution so i can't speak for how well cisco phones work there in comparison to any other brand of phones |
| 13:40.26 | leifmadsen | I've used Cisco phones -- they are very out of date, firmware wise compared to any Polycom solution |
| 13:40.37 | leifmadsen | if you're using anything with the XML configs on the Cisco phones -- good luck to ya |
| 13:41.09 | *** join/#asterisk jicksta (n=jicksta@c-98-210-108-28.hsd1.ca.comcast.net) |
| 13:41.09 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 13:41.32 | *** join/#asterisk propellerhead (n=yogurt2u@host111.190-31-111.telecom.net.ar) |
| 13:41.43 | lowtek | Naikrovek: The 7940/7960 work *good* but only do a single G.729 channel at a time. The Polycom's work *great* and frankly have taken everything we've thrown at them. |
| 13:42.01 | [TK]D-Fender | leifmadsen: Well the SIP device did seem to abort... not so much detail whiy.. |
| 13:42.04 | leifmadsen | we're still talking about this? :) Just use Polycom and live a happy life. |
| 13:42.10 | leifmadsen | [TK]D-Fender: aye... |
| 13:42.59 | lowtek | Naikrovek: If there was a huge investment in 7940's (we see that a lot) then the transition to asterisk might need to be with the Cisco phones. There are just sooooooo many 7940's out there. |
| 13:43.01 | Naikrovek | anyone know what kind of thing happens if you have phones configured to use G729 in a conference together without a G729 license on the * server? |
| 13:43.22 | *** part/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 13:43.42 | Naikrovek | had to ship all his testing phones away to India for that office |
| 13:44.05 | [TK]D-Fender | Naikrovek: DOA <- |
| 13:44.22 | Naikrovek | calls drop or just silence |
| 13:44.24 | shido6 | next time test on your dev platform before doing that |
| 13:44.24 | mort_gib | Naikrovek: allow more than one codecs, but apart from that I agree with TK |
| 13:44.38 | [TK]D-Fender | Naikrovek: Every call has to transcode back, and that is a large cumulative load factor as well. |
| 13:44.53 | [TK]D-Fender | Naikrovek: Drop. Like a ROCK |
| 13:44.59 | Naikrovek | harsh |
| 13:45.01 | Naikrovek | thansk |
| 13:45.05 | Naikrovek | thanks, even |
| 13:46.07 | Naikrovek | i bought a polycom for testing with at home but i have nothign here to test with anymore. |
| 13:47.08 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 13:47.49 | michael-i | new objdump output on my missing loadable module symbol problem (http://pastebin.ca/1533823) here, res_monitor loads, app_dial does not. Any input is welcome |
| 13:50.27 | leifmadsen | [TK]D-Fender: well, the RTP is definitely stopping, but that should be expected since I'm on hold (I'm on the DAHDI channel), and there would be no audio flowing from the SIP channel (since they placed me on hold) |
| 13:50.35 | leifmadsen | continues to review more debugging logs |
| 13:50.37 | *** join/#asterisk brah (n=asdfaf@86-126-16-190.fibertel.com.ar) |
| 13:50.48 | [TK]D-Fender | leifmadsen: Check and see if the BYE came in with a reason |
| 13:50.55 | leifmadsen | okie |
| 13:52.24 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 13:52.27 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 13:53.05 | leifmadsen | [TK]D-Fender: hmmm... this is odd: Reason: SIP;description="User Hung Up"^M |
| 13:53.09 | lowtek | Polycom experts: Is there a way, with EFK or another function, to program a "Page" button that will let you hit it, then another contact key to effectively build the dial-string before sending it? I can get the "Page" button to work, but whenever you hit a contact afterworks it just picks up a line and dials the contact. |
| 13:53.21 | [TK]D-Fender | leifmadsen: Who is the user? |
| 13:53.21 | leifmadsen | I wonder if this has something to do with the softphone being used, and nothing to do with Asterisk.... |
| 13:53.25 | leifmadsen | [TK]D-Fender: eyeBeam |
| 13:53.27 | [TK]D-Fender | leifmadsen: Sounds blatant... |
| 13:53.40 | [TK]D-Fender | leifmadsen: No, I mean the PEBKAC ;) |
| 13:54.02 | leifmadsen | [TK]D-Fender: entirely possible, but it's exactly 30 seconds every time, so it has to be the software |
| 13:54.20 | [TK]D-Fender | leifmadsen: Trial version timeout? :) |
| 13:54.36 | leifmadsen | [TK]D-Fender: maybe.... but that'd be pretty dumb :) |
| 13:55.02 | [TK]D-Fender | leifmadsen: par for the course... I'd test another as a sanity check |
| 13:55.11 | leifmadsen | [TK]D-Fender: yep, gonna do that right now |
| 13:55.13 | [TK]D-Fender | leifmadsen: X-Lite would be a logical choice |
| 13:58.44 | leifmadsen | [TK]D-Fender: well, he's gonna check some settings, because this is apparently a registered version of eyeBeam |
| 13:59.13 | leifmadsen | but from what I can tell in the logging, it is the softphone that is doing the hanging up |
| 14:00.56 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:00.56 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:00.58 | [TK]D-Fender | leifmadsen: Perhaps an app they use uses a key that is bound to eyeBeam as disconnect regardless of focus |
| 14:02.01 | *** join/#asterisk larin (n=larin@mail.cs-service.by) |
| 14:02.10 | *** join/#asterisk DarkRift (n=dark@65.92.166.33) |
| 14:02.24 | leifmadsen | doesn't sound all that likely as they wouldn't have developed anything like that yet |
| 14:02.50 | leifmadsen | since this is a brand new install |
| 14:03.00 | manxpower | 30 seconds is the magical "don't have canreinvite=no, but need that option" symptom isn't it? |
| 14:03.31 | manxpower | oh, leifmadsen is having the problem. nevermind, he knows what he is doing. |
| 14:03.41 | *** join/#asterisk oej (n=olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 14:03.47 | leifmadsen | manxpower: heh, and I have canreinvite=no :) |
| 14:04.08 | leifmadsen | manxpower: from what I can tell in the logs, the eyeBeam is hanging up the call after 30 seconds after placing it on hold for some reason |
| 14:06.39 | fenlander | 30s also shows up if there's a missing ACK, usually dropped by some broken proxy |
| 14:06.49 | *** join/#asterisk coppice (n=chatzill@52.204.17.210.dyn.pacific.net.hk) |
| 14:08.04 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 14:08.12 | manxpower | leifmadsen: is VAD enabled, maybe ann RTP timeout is happening? |
| 14:09.06 | [TK]D-Fender | manxpower: He's on hold <- |
| 14:09.18 | [TK]D-Fender | manxpower: and its eyebeam calling for it and giving up |
| 14:12.54 | *** join/#asterisk hajvan (i=hajvan@avlianer.hajvan.net) |
| 14:12.59 | hajvan | greets |
| 14:13.30 | *** join/#asterisk rene- (n=renemend@200.34.66.137) |
| 14:13.35 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733667.dsl.bell.ca) |
| 14:15.46 | *** join/#asterisk jcape (n=jcape@209.120.251.81) |
| 14:15.48 | hajvan | anyone who can give me a hint :)? Dialogic Corporation Diva BRI-2FX PCI as a hardware, asterisk-1.4.26.1 installed prom source ( asterisk-1.4.26.1, asterisk-addons-1.4.9, dahdi-linux-2.2.0.2, dahdi-tools-2.2.0, libpri-1.4.10.1) Diva4Linux_installer_9.0-109-82.bin Kernel module and Diva tools installd and working, everything fine just my asterisk isn't answering the call on MSN i set up (Germany ISDN Line) |
| 14:17.13 | leifmadsen | [TK]D-Fender: it was an eyeBeam option that is enabled by default, which is absolutely retarded |
| 14:18.35 | leifmadsen | [TK]D-Fender: http://forums.counterpath.com/viewtopic.php?f=1&t=14922&p=54993&hilit=hangup+30+seconds#p54993 |
| 14:22.01 | leifmadsen | [TK]D-Fender: and this link: https://support.counterpath.com/default.asp?W30 |
| 14:22.07 | leifmadsen | awesome eyeBeam! |
| 14:22.21 | *** join/#asterisk cusco_ (n=tralala@2001:0:53aa:64c:1cea:d77c:2ac0:762d) |
| 14:22.24 | cusco_ | hi |
| 14:22.50 | fenlander | leifmadsen: is eyebeam even dropping the calls if rtcp is still flowing? |
| 14:22.58 | GeminiDomino | Anyone have any hints on making WaitForSilence a bit less sensitive? |
| 14:23.04 | leifmadsen | fenlander: that was the problem -- it was dropping the calls |
| 14:23.59 | fenlander | leifmadsen: that's badly broken, rtcp reports should keep it alive even if no rtp. And don't get me started on sip session timers ;) |
| 14:24.14 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:24.22 | leifmadsen | fenlander: I don't care about RTCP, so regardless, it is now disabled :) |
| 14:26.01 | hershel | i have a zapata-channels.conf file but no zapata-auto.conf I had genzapconf make the files by itself. In * prompt I see no zap channels. Do I need to copy the zapata-channels.conf to zapata-auto.conf ? |
| 14:27.09 | cusco_ | asterisk is returning 448, codec not accepted |
| 14:28.06 | manxpower | cusco_: allow the codec than |
| 14:28.18 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:28.37 | *** join/#asterisk InfoNutz (n=infonutz@204.50.209.225) |
| 14:28.56 | tzafrir_laptop | hershel, technically you don't *need* those files. You can either #include any of them into zapata.conf or copy its content to the end of zapata.conf |
| 14:29.43 | tzafrir_laptop | hajvan, this card uses chan_capi, IIRC |
| 14:30.18 | tzafrir_laptop | (or its own version of chan_capi) |
| 14:30.28 | tzafrir_laptop | It does not use chan_dahdi |
| 14:30.45 | [TK]D-Fender | leifmadsen: Sofa king wheat hearted..... |
| 14:30.52 | hajvan | tzafrir_laptop: sure i have installed chan_capi-1.1.4 |
| 14:31.45 | tzafrir_laptop | hajvan, well, that's about as much as I know about chan_capi |
| 14:31.48 | hajvan | tzafrir_laptop: i can make a test call and a test fax by using Diva test tools |
| 14:32.04 | hajvan | tzafrir_laptop: hmm ... |
| 14:34.12 | hershel | tzafrir_laptop: OK, I hear. but I did copy zapata-channels.conf to zapata-auto.conf and then restarted asterisk and zaptel. still no channels showing. but when I restart zaptel it doesn't start/restart like a regular service. it just says Zaptel telephone kernel driver: zaptel |
| 14:34.54 | manxpower | hershel: zaptel will start with no errors even if you don't have any cards in the system. listen to tzafrir_laptop |
| 14:35.06 | bmoraca | hershel: then you didn't hear him very well. you need to put that information in zaptel.conf or include one of those two files in zaptel.conf. it doesn't do that automatically. |
| 14:35.33 | hershel | ah, i didn't realize it doesn't pick up automatically. i thought that was an OPTION. OK. |
| 14:35.46 | hajvan | tzafrir_laptop: please tell me if i'm wrong, capi.conf setup (ntmode=no,isdnmode=msn,incomingmsn=XXXXXX,context=foobar), extensions.conf setup [foobar] \ include => demo should give me a answer as i give a call? |
| 14:36.42 | tzafrir_laptop | hajvan, you're probably wrong to assume I know how chan_capi works :-( |
| 14:37.02 | hajvan | hmm |
| 14:37.21 | bmoraca | but but but...you're answering questions...you must know how EVERYTHING works! |
| 14:38.29 | hajvan | tzafrir_laptop: no big deal, :) i just want to chek if my "asterisk logic" is ok |
| 14:39.04 | *** join/#asterisk Weezey (n=ohno@lan3.LO.iasl.com) |
| 14:39.52 | hershel | tzafrir_laptop KNOWS what he's talking about. he just fixed my setup and I got ztdummy via a post on a list he made in 2006. Googled and found it. :) |
| 14:40.41 | *** part/#asterisk cobra2599 (n=cobra259@cpe-66-25-60-236.tx.res.rr.com) |
| 14:40.58 | skrusty | can you use playback to play an audio file in a sub directory of your sounds directory? |
| 14:41.30 | skrusty | i.e. {defaultaudiodir}/subdir/myprompt |
| 14:41.56 | tamiel | skrusty: yes |
| 14:41.56 | tzafrir_laptop | skrusty, yes |
| 14:42.24 | tzafrir_laptop | hershel, what's the output of: cat /proc/zaptel/* |
| 14:42.44 | skrusty | cheers guys :) |
| 14:43.18 | hershel | Pardon me for this, gentleman, but I just called my box AND I HEARD THE CONGRATULATIONS MESSAGE!! |
| 14:44.07 | hershel | i will give anyone the # if u want to test also. LOL!! |
| 14:44.31 | [TK]D-Fender | hershel: No, we'll take your word for it... |
| 14:44.55 | hershel | lol. sorry. i was told it would take DAYS to set this up. all i need now is to connect to agi. and i just started today. |
| 14:45.22 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:45.22 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:46.10 | hershel | <PROTECTED> |
| 14:46.12 | hershel | looks good to me I think |
| 14:46.46 | hershel | except for misses I guess. |
| 14:47.33 | cusco_ | manxpower: it is suposed to be allowed, how can I check |
| 14:48.50 | *** join/#asterisk teknomega (n=cmr@75-147-97-177-Philadelphia.hfc.comcastbusiness.net) |
| 14:48.54 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 14:49.53 | cusco_ | how can I force a channel to be accepted |
| 14:49.54 | cusco_ | ? |
| 14:50.48 | *** join/#asterisk Ex_peter (n=Ex_peter@unaffiliated/expeter/x-019426) |
| 14:51.34 | [TK]D-Fender | cusco_: How the hell is a channel "proposed"? |
| 14:51.56 | cusco_ | by the sip protocoll |
| 14:52.12 | [TK]D-Fender | cusco_: You can't force acceptance. they have to agree |
| 14:52.12 | cusco_ | Global Signalling Settings: |
| 14:52.12 | cusco_ | --------------------------- Codecs: 0x10e (gsm|ulaw|alaw|g729) |
| 14:52.17 | cusco_ | ti is allowed |
| 14:52.20 | cusco_ | ok |
| 14:52.25 | cusco_ | why can't asterisk agree? |
| 14:52.28 | [TK]D-Fender | cusco_: And you aren't showing us the FAILURE. |
| 14:52.33 | [TK]D-Fender | cusco_: PAStebIN is your friend. |
| 14:52.34 | cusco_ | ok hold |
| 14:52.35 | [TK]D-Fender | ~pb |
| 14:52.35 | 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://bin.cakephp.org/ , or apt-get install pastebinit |
| 14:52.37 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 14:52.39 | cusco_ | ok |
| 14:53.49 | cusco_ | http://paste.debian.net/44501/ |
| 14:53.52 | cusco_ | there |
| 14:55.50 | [TK]D-Fender | cusco_: SIP/2.0 488 Not acceptable here <- incompatible codecs |
| 14:56.06 | [TK]D-Fender | cusco_: I see G.729 in the list. Do yuo have licenses installed? |
| 14:56.53 | [TK]D-Fender | cusco_: And the meida is listed as SRTP which last I checked * does not support |
| 14:56.58 | [TK]D-Fender | media* |
| 14:59.09 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:00.01 | cusco_ | is that the softphone configuration? |
| 15:02.38 | cusco_ | what is the media srtp ? |
| 15:04.25 | *** part/#asterisk larin (n=larin@mail.cs-service.by) |
| 15:04.31 | cusco_ | ok secure rtp |
| 15:04.38 | cusco_ | you are right thats it |
| 15:04.41 | cusco_ | thank you [TK]D-Fender |
| 15:05.14 | hershel | I found here http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf that "If you are in Israel, the following is important: " but it seems to be regarding a ISDN PRI Switch Configuration whcih I don't think I have. |
| 15:05.23 | hershel | But I do believe I am in Israel. ;) |
| 15:09.08 | *** join/#asterisk bluOxigen (n=xainix20@static-host119-73-66-108.link.net.pk) |
| 15:15.03 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.83) |
| 15:17.12 | [TK]D-Fender | cusco_: You're welcome |
| 15:17.18 | [TK]D-Fender | NEXT!!@@!@! (c) BKW |
| 15:18.00 | hershel | Sorry for this AGAIN, but I just connected the sample php cgi given here: http://www.voip-info.org/wiki/view/Asterisk+AGI+php AND IT WORKS!! Do u know that on asterisk's form (digium I guess) they said this would take me DAYS. |
| 15:20.27 | *** join/#asterisk dajhorn (n=dajhorn@206.16.96.160) |
| 15:20.51 | skrusty | hershel: cool :) |
| 15:21.11 | skrusty | AGIs are really nice and easy to use |
| 15:22.02 | skrusty | just connected my touch hd to asterisk using SipConfig.cab :D |
| 15:22.08 | cusco_ | AGI? |
| 15:22.38 | skrusty | Asterisk Gateway Interface |
| 15:22.58 | skrusty | allows a script/exe etc to take control of call handling |
| 15:24.14 | hershel | yes, cool. 3 hours WITH a good bit of help here. and now to write the script. that part I know how to do . ;) |
| 15:24.20 | hershel | thanks to all who helped me. |
| 15:24.28 | cusco_ | hmm... ok.. |
| 15:24.41 | cusco_ | is that for some click to talk script? |
| 15:26.02 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 15:29.30 | *** join/#asterisk rossand (n=rossand@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 15:29.45 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:33.31 | *** join/#asterisk Ex_peter (n=Ex_peter@unaffiliated/expeter/x-019426) |
| 15:35.04 | [TK]D-Fender | hershel: DAYS is what it takes to learn about so much more like setting up SIP phones, dial plans, etc |
| 15:36.33 | hershel | [TK]D-Fender: I don't doubt it. but i explained to them EXACTLY what i want to do. :) |
| 15:37.05 | hershel | cusco: AGI allows * to answer the phone and then send full control of the call to a script (or EXE I guess) so the script can receive the numbers typed and send back sound. |
| 15:37.14 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 15:37.25 | hershel | so in 3 hours i got it setup and now ALL of my logic will simply be in my PHP script. |
| 15:37.45 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 15:38.43 | [TK]D-Fender | hershel: Estimates also depend on the person, and who is helping them |
| 15:39.11 | hershel | This is a very good point. without this channel i would NEVER have gotten this in a mere 3 hours |
| 15:40.52 | rob0 | For some folks, seeking assistance in IRC is a net negative. |
| 15:41.08 | hershel | net negative ? |
| 15:41.14 | Qwell | hershel: some of us get stuck here. |
| 15:41.46 | hershel | aha. I understand. Either I got lucky or my problems were easy. :) |
| 15:41.54 | rob0 | I mean, they are wasting their time as well as the time of those who try to help. |
| 15:45.42 | *** part/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 15:46.35 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:47.34 | *** join/#asterisk scunizi (n=scunizi@adsl-99-40-45-60.dsl.sndg02.sbcglobal.net) |
| 15:48.20 | hershel | I found here http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf that "If you are in Israel, the following is important: " but it seems to be regarding a ISDN PRI Switch Configuration whcih I don't think I have. |
| 15:49.34 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:54.44 | *** join/#asterisk semaries (n=martin@stgt-5d84918a.pool.einsundeins.de) |
| 15:54.57 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 15:55.35 | garymc | i know i should be elsewhere but nobody there right now - I just installed AsteriskNOW on my server and after everything is installed im at a black promt screen. Is this how its supposed to be? I thought it might have been a graphical thing so I could easily see how stuff was working etc. |
| 15:55.58 | garymc | now im installed i dont know what to do |
| 15:56.18 | Qwell | garymc: open a browser, point it to the IP of the server |
| 15:56.33 | garymc | hwo do i know the ip? |
| 15:56.52 | garymc | should my server now be plugged into my ethernet switch? |
| 15:57.13 | bmoraca | yikes |
| 15:58.09 | rob0 | senses an upcoming "net negative" |
| 15:58.28 | garymc | :S |
| 15:58.53 | Qwell | garymc: read the quickstart guide on asterisknow.org |
| 15:58.56 | heedly | hhehehe |
| 15:59.05 | garymc | im looking lol |
| 16:00.10 | manxpower | ~asterisknow |
| 16:00.11 | infobot | from memory, asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 16:00.42 | garymc | thankyou info |
| 16:00.45 | garymc | bot |
| 16:01.07 | garymc | Qwell the link on there site just reloads the same page? |
| 16:01.45 | garymc | found another link |
| 16:02.11 | *** join/#asterisk mattboll (n=mattboll@78.238.188.24) |
| 16:02.14 | mattboll | hi |
| 16:03.49 | WHYS | i'm looking for a way to timeout a DIAL if the peer can not be reached and the call not setup. right now it seems to block for several minutes trying to establisha connection. |
| 16:05.17 | lowtek | WHYS: Dial(tech,30) ; 30 seconds |
| 16:05.21 | *** join/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 16:05.49 | *** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net) |
| 16:06.10 | WHYS | lowtek: that's only after a connection is setup and the call not answered. I need something if the peer is off line. |
| 16:06.19 | kaldemar | WHYS: what kind of peer? |
| 16:06.31 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 16:06.36 | WHYS | Unified messaging. |
| 16:06.39 | *** part/#asterisk sjobeck (n=Adium@208-151-246-203.dq1sn.easystreet.com) |
| 16:07.04 | manxpower | WHYS: there is no way to do that. Use a decent service. qualify=somenumber will make a peer unreachable after a non-response to being polled for somenumber of seconds, but that has nothing to do with Dial |
| 16:07.24 | leifmadsen | Asterisk doesn't call IAX or SIP peers if they are not reachable by default (i.e. qualify=200) |
| 16:07.30 | Naikrovek | 2000 |
| 16:08.03 | mattboll | WHYS: did you look at ${DIALSTATUS} ? may be you can find something |
| 16:08.06 | manxpower | that does NOT say poll every 2000ms, that says when polling if there is no response for 2000ms mark as unreachable |
| 16:08.14 | garymc | yeah it says a url to the GUI will be displayed. but it isnt? |
| 16:08.29 | kaldemar | autokill is nice with IAX2 |
| 16:08.46 | WHYS | Right. I still have to do something because UM listens on only two different ports, and switches randomly. since teh firewall is stealth I don't even get a response back and everything blocks. |
| 16:08.48 | Naikrovek | manxpower: how often does it poll, and can you force a poll |
| 16:09.13 | WHYS | I turned qualify off so it's doesn't check up front - same problem there. |
| 16:09.17 | manxpower | Naikrovek: use the source luke (every 30 seconds, I think, but look at the source), no way to force ap poll |
| 16:09.27 | Naikrovek | k |
| 16:09.45 | manxpower | WHYS: It sucks to be you, doesn't it. |
| 16:09.47 | kaldemar | Naikrovek: qualifyfreq |
| 16:09.50 | WHYS | :( |
| 16:09.59 | Naikrovek | kaldemar: thanks |
| 16:10.03 | manxpower | WHYS: it would be impossible for a SIP server to change it's SIP ports or clients would not connect. |
| 16:10.05 | kaldemar | Naikrovek: see the sample sip.conf |
| 16:10.12 | manxpower | kaldemar: that must be a 1.4ism |
| 16:10.16 | Naikrovek | ok |
| 16:10.20 | WHYS | It's not like I WANT to use a microsoft product. It's an employer thing. I have to eat. |
| 16:10.23 | kaldemar | manxpower: no, it's in 1.6 |
| 16:10.34 | manxpower | kaldemar: Cool. It SHOULD have been in 1.4 |
| 16:10.45 | manxpower | kaldemar: remember to mention the version when talking about new options |
| 16:11.15 | WHYS | still, is is proper to * to simply block if a peer goes off line? that's not graceful |
| 16:11.19 | kaldemar | well.. let's see about the new... |
| 16:11.38 | tripps | anyone recommend any good sip providers out there with 24/7 support, SLAs, etc., but doesn't price their services by "sip trunks" like bandwidth.com? or at least has them dirt cheap? |
| 16:12.18 | leifmadsen | WHYS: chan_sip basically works by sending an INVITE at 1, 2, 4, 8, 16 second intervals (if I remember correctly). You can change the source pretty simply to not try 6 times |
| 16:12.40 | leifmadsen | tripps: wait, you want cheap, but 24/7 support and SLAs? nice... |
| 16:12.47 | leifmadsen | does not compute. |
| 16:13.09 | leifmadsen | goes to lunch! Stuffed peppers ftw |
| 16:13.23 | kaldemar | qualifyfreq came with 1.6.0. so it's been around for a while now. but you have a point, manxpower. |
| 16:13.24 | tripps | leifmadsen, let me clarify. not cheap. just not an anachronistic business model |
| 16:13.43 | manxpower | kaldemar: do not underestimate 1.4 |
| 16:13.46 | WHYS | Sure, I could change the source. Nice to know this one might be easier, but I'm not confortable there. sigh. Maybe I could add an option. |
| 16:13.58 | *** part/#asterisk semaries (n=martin@stgt-5d84918a.pool.einsundeins.de) |
| 16:14.02 | manxpower | WHYS: you are welcome to get a refund and use a different product. |
| 16:14.20 | kaldemar | manxpower: i don't, my guess would be that it's still the most used branch. |
| 16:14.44 | bmoraca | WHYS: what happens in the dialplan in asterisk is entirely up to you. if a peer isn't reachable, you can make the dialplan behave almost anyway you want. |
| 16:14.57 | WHYS | Thanks, I'll take you up on that. I spent $500 to give digium some cash and get a little support now and then. manxpower are you signing the check? |
| 16:15.07 | manxpower | There are two kinds of people that use Asterisk. There are the people that fight Asterisk's oddities and limitations -- those people live miserable pointless lives. There are also the people that accept Asterisk's oddities and limitations and work with them -- those people live happy, joyful lives. |
| 16:15.13 | tripps | SIP carriers should have burstability (number of concurrent calls) or 95th percentile type models like IP services. If my usage of 15,000 minutes/month is 99.999% 1-2 concurrent calls, but during peak periods I may have 10 concurrent calls, I shouldn't have to purchase 10 sip trunks at $25 each that I'm not going to use. it's ridiculous. |
| 16:15.49 | *** part/#asterisk hershel (n=hershel@213.8.21.65) |
| 16:15.50 | WHYS | I'l love to be happy all the time. I would rather not fight, but I still need a solution. |
| 16:16.15 | *** join/#asterisk fnord0 (n=fnord0@unaffiliated/fnord0) |
| 16:16.29 | WHYS | Maybe I'll have to contribute. It's about time I guess. |
| 16:16.52 | WHYS | Lets see... is * written in perl. :) |
| 16:17.27 | errr | yes, but they compiled the perl to make it look more likec :P |
| 16:17.43 | errr | like c |
| 16:19.48 | outtolunc | if you have 10 cars, but only use 1-2 usually, you shouldn't have to register the others right? |
| 16:20.05 | WHYS | If you don't drive them. |
| 16:20.30 | outtolunc | he wants to drive 1-2, and the others 'when he feels like it' |
| 16:20.32 | bmoraca | the key word is "usually". if you PNO a car, you CAN'T legally drive it. |
| 16:20.35 | outtolunc | but only pay for 1-2 |
| 16:22.17 | *** join/#asterisk citywok (n=chatzill@67.148.102.15) |
| 16:22.43 | *** join/#asterisk freddyk (n=freddy@82.55.140.59) |
| 16:23.18 | *** join/#asterisk bluOxigen (n=xainix20@static-host119-73-71-53.link.net.pk) |
| 16:26.58 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 16:27.04 | fcois93 | hello all |
| 16:27.43 | fcois93 | is it possible to lock a Conference when nobody is online to be shure that the conference is offline? |
| 16:28.48 | *** join/#asterisk ebroad (n=elazar@72.11.213.194) |
| 16:29.39 | ebroad | hello |
| 16:30.29 | *** join/#asterisk bluOxigen (n=xainix20@static-host119-73-71-53.link.net.pk) |
| 16:30.49 | ebroad | i am experimenting with t.38 on asterisk-svn-trunk-r212672 |
| 16:31.33 | ebroad | and asterisk keeps responding with 488 not acceptable here after the reinvite |
| 16:31.47 | ebroad | when using zoiper as a client |
| 16:32.56 | ebroad | sip.conf is pretty simple, faxdetect=yes under general |
| 16:33.05 | ebroad | and 2 sip friends |
| 16:33.16 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 16:33.25 | ebroad | the fax extension is defined in extensions.ael |
| 16:33.36 | ebroad | anybody? |
| 16:33.58 | *** join/#asterisk ArchGT (n=ArchGT@unaffiliated/archgt) |
| 16:35.09 | manxpower | ebroad: faxdetect is a zaptel only feature |
| 16:36.27 | ebroad | manxpower: its listed in sip.conf under T.38 passthrough |
| 16:36.37 | leifmadsen | ebroad: what version? |
| 16:37.01 | leifmadsen | ebroad: there have been several T.38 changes recently, so you may wish to try a recent checkout from a branch |
| 16:37.13 | ebroad | see https://reviewboard.asterisk.org/r/69/ |
| 16:38.39 | ebroad | leifmadsen, just checked out last night |
| 16:39.13 | ebroad | leifmadsen, r212672 |
| 16:39.25 | *** join/#asterisk Defraz (n=T0tal@corp.fuzecore.com) |
| 16:40.12 | *** join/#asterisk LiNeTuX (n=LiNeTuX@fl-209-26-240-156.sta.embarqhsd.net) |
| 16:47.02 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 16:47.54 | manxpower | ebroad: try #asterisk-dev |
| 16:48.58 | ebroad | manxpower, thanx |
| 16:50.46 | *** join/#asterisk pryorda (n=dpryor@unaffiliated/irated) |
| 16:51.09 | pryorda | need some help setting this up trixbox people dont have any ideas so im thinking maybe this channel wil |
| 16:51.20 | pryorda | I can not get any outbound calls to work with les.net |
| 16:51.27 | pryorda | anyone else worked with les.net |
| 16:51.27 | pryorda | ? |
| 16:52.20 | manxpower | ~trixbox |
| 16:52.21 | infobot | methinks trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 16:52.48 | *** join/#asterisk bbryant (n=brett@c-68-59-22-114.hsd1.sc.comcast.net) |
| 16:52.58 | bmoraca | ~elastix |
| 16:52.59 | infobot | elastix is, like, a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes Elastix VERY difficult to support, and is not supported in #asterisk. Try asking in #Elastix or on their forums at http://www.elastix.org |
| 16:53.17 | bmoraca | interesting |
| 16:54.31 | pryorda | Let me rephrase then |
| 16:54.31 | *** part/#asterisk ebroad (n=elazar@72.11.213.194) |
| 16:55.02 | pryorda | asterisk on trixbox is not working correctly to connect to les.net trunk |
| 16:55.03 | pryorda | :) |
| 16:55.06 | pryorda | that better |
| 16:55.57 | leifmadsen | it's still trixbox controlling asterisk, and this channel is for asterisk vanilla support |
| 16:56.01 | pryorda | I understand that you guys do not support trixbox here. Im asking has anyone else had issues with the default configs from les .net |
| 16:57.16 | *** join/#asterisk rossand (n=rossand@99.246.183.44) |
| 16:57.37 | bmoraca | pryorda: your asterisk configuration is handled by freepbx or pbxconfig or whatever they call it now...your issue is likely how you input your configuration into that front-end, and that it's probably incorrect. |
| 16:58.38 | [TK]D-Fender | "Trunk config", etc |
| 16:58.44 | [TK]D-Fender | "Inbound route". |
| 16:58.49 | [TK]D-Fender | What utter garbage... |
| 16:58.56 | pryorda | [TK]D-Fender: ? |
| 16:58.58 | KyleK | "Junk for our trunk" |
| 16:59.24 | pryorda | Thanks for your help guys.. |
| 16:59.29 | KyleK | pryorda: he's giving you hints for what to look for in the interface |
| 16:59.35 | [TK]D-Fender | that too |
| 16:59.40 | pryorda | I will look more into it |
| 16:59.42 | bmoraca | lol |
| 16:59.52 | pryorda | inbound works fine :) |
| 16:59.58 | pryorda | just outbound |
| 17:00.02 | KyleK | pryorda: the default configs going out work for me |
| 17:00.06 | [TK]D-Fender | pryorda: those GUI's invent so many other terms and places to go to set up various bits, and all we'll hear in here is "It doesn't work". |
| 17:00.29 | [TK]D-Fender | pryorda: See when you control *, you see everything right at the start and there are only 2 places to look. Not so with GUI's in the way |
| 17:00.29 | KyleK | pryorda: my only complaint about les.net is the lack of https :) |
| 17:00.44 | [TK]D-Fender | KyleK: For the payment page? |
| 17:00.58 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:01.14 | KyleK | well they accept paypal for payment, but I'd like to login to the interface to at least be https |
| 17:01.46 | [TK]D-Fender | KyleK: Yeah, it is off-putting |
| 17:02.32 | bmoraca | but it's an extra $25/year for an SSL cert! |
| 17:03.36 | InfoNutz | hello, how do i setup the dialplan so that when a user gets a voicemail prompt afer dialing an extension to leave a message they can press * to access their admin section instead of dialing another extension to access voicemailMain()? |
| 17:04.25 | bmoraca | look up the 'a' extension |
| 17:04.35 | [TK]D-Fender | ^^^ |
| 17:04.54 | [TK]D-Fender | InfoNutz: go read up on ALL of the "Asterisk Standard Extensions" |
| 17:05.04 | atis_work | zmb: generally it's res_mysql.conf |
| 17:05.11 | KyleK | i guess self signed would look worse to customers but it'd make me more willing to log in and change stuff on other peoples wireless |
| 17:05.36 | InfoNutz | thanks! i'll go over them again, musta missed it |
| 17:05.51 | bmoraca | InfoNutz: http://www.voip-info.org/wiki/view/Asterisk+a+extension |
| 17:07.13 | pryorda | KyleK: k thanks |
| 17:09.25 | bmoraca | has anyone in here successfully integrated a Lucent/Ascend MAX TNT with asterisk? |
| 17:15.52 | skrusty | evening |
| 17:16.14 | raden_work | [TK]D-Fender, i updated firmware on FVX538 last night now i cant even register any ideas ? |
| 17:16.26 | raden_work | [Aug 18 12:13:44] NOTICE[10630]: chan_sip.c:9489 sip_reg_timeout: -- Registration for '17772445766@callcentric.com' timed out, trying again (Attempt #7) |
| 17:16.36 | [TK]D-Fender | raden_work: PB the entire attempt |
| 17:16.43 | [TK]D-Fender | raden |
| 17:16.57 | raden_work | yes ? |
| 17:17.08 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:17.51 | [TK]D-Fender | raden_work: leftovers. get the PB |
| 17:18.13 | raden_work | is there a way to get more info on it trying to register than just the CLI output ? |
| 17:18.18 | [TK]D-Fender | raden_work: If you updated your firmware, double-check your forwarding <- |
| 17:18.25 | [TK]D-Fender | radeSIP DEBUG. |
| 17:18.30 | [TK]D-Fender | raden_work: SIP DEBUG. |
| 17:18.32 | [TK]D-Fender | gah |
| 17:18.37 | [TK]D-Fender | raden_work: You know this already |
| 17:18.47 | raden_work | just not showing alot one moment |
| 17:19.12 | [TK]D-Fender | radeI'm sure its showing ENOUGH |
| 17:19.27 | [TK]D-Fender | raden_work: And go verify your rules are all as expected. |
| 17:19.57 | freddyk | is anyone using zaphfc over trunk ? |
| 17:20.37 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 17:21.16 | raden_work | http://pastebin.com/d4cffe3bb |
| 17:22.26 | [TK]D-Fender | raden_work: Check your firwarding, and is your WAN IP the same? |
| 17:22.42 | raden_work | yes wan ip static double checking all settings and forwarding |
| 17:23.01 | KyleK | are you still fighting with wifi phones? |
| 17:23.11 | raden_work | WAN #1 IP Address: 69.179.99.17 |
| 17:23.13 | [TK]D-Fender | KyleK: No, we fixed that last night |
| 17:23.45 | raden_work | ok one step at a time lets get registered here |
| 17:24.12 | raden_work | is this a valid subnet mask ? IP Address: 69.179.99.17 |
| 17:24.12 | raden_work | Subnet Mask: 255.255.255.255 |
| 17:24.18 | drmessano | ROFL |
| 17:24.24 | drmessano | newp |
| 17:24.29 | lowtek | raden_work: No. |
| 17:24.43 | raden_work | thats whats being assigned by our ISP |
| 17:24.49 | drmessano | I_HAVE_NO_NEIGHBORS_FAIL |
| 17:24.55 | lowtek | raden_work: You're sure that's not your broadcast address? |
| 17:24.56 | drmessano | Sounds like you're wall gardened |
| 17:25.00 | raden_work | actually i should not say that its what the router doing on its own |
| 17:25.07 | raden_work | drmessano, ? |
| 17:26.24 | drmessano | http://en.wikipedia.org/wiki/Walled_garden_(technology) |
| 17:26.33 | drmessano | They have you in a tiny box |
| 17:28.08 | *** join/#asterisk raden (n=tanning@69-179-99-17.stat.centurytel.net) |
| 17:28.36 | bmoraca | that could be a valid subnet depending on the configuration and type of router. |
| 17:28.54 | bmoraca | some pppoe static services will end up giving you a /32 |
| 17:28.54 | raden | http://pastebin.com/d124bfc9e |
| 17:29.00 | *** join/#asterisk stupidnic (n=foo@cpe-70-94-229-122.sw.res.rr.com) |
| 17:29.00 | raden | ok |
| 17:29.09 | raden | [TK]D-Fender, any input ? |
| 17:30.00 | [TK]D-Fender | raden: Drop a NIC into your server and run it direct |
| 17:30.14 | [TK]D-Fender | raden: Or replace that POS with --whatever-- |
| 17:30.19 | drmessano | bmoraca: Looks like the gateway is on a different subnet, so thats about right |
| 17:30.39 | drmessano | Shame you dont have a Linksys router laying around you could use |
| 17:30.41 | raden | [TK]D-Fender, server has 2 nics |
| 17:30.52 | drmessano | ZIING |
| 17:30.55 | [TK]D-Fender | radenGo use it then |
| 17:31.11 | raden | drmessano, i have another linksys wrt54gl and a buffalo airstation with DDwrt here |
| 17:31.19 | stupidnic | I need some assistance in troubleshooting a TDM-400. For reasons I can't determine it occasionally drops station connections and I am unable to get a dial tone on stations until I issue a reload in the asterisk console. Even after I do that it still can't pick up on the FXO channel. |
| 17:31.29 | bmoraca | drmessano: i've stopped trying to understand why some static DSL providers do things and number things the way they do...but that's not the first time i've seen numbering like that |
| 17:31.47 | raden | [TK]D-Fender, you thinking its the netgear giving all these problems |
| 17:31.55 | [TK]D-Fender | raden: Yes |
| 17:32.17 | raden | ok lemme finish checking the settings in the router and ill get back |
| 17:32.18 | bmoraca | raden: what kind of netgear? RangeMAX V3s do not have a SIP passthrough mode...the V2s do, though. |
| 17:32.25 | raden | FVX538 |
| 17:32.55 | bmoraca | make sure SPI is turned off |
| 17:32.56 | raden | FVX538 v1.1 |
| 17:33.04 | raden | looking for it |
| 17:34.22 | bmoraca | it does aparently have a SIP ALG, at least according to specs. i'd recommend turning both the SIP ALG and the SPI firewall off. depending on what they actually mean by "SIP ALG", though, that might need to stay on. either way, SPI needs to be off. |
| 17:34.38 | drmessano | HAHAHAH |
| 17:34.50 | drmessano | I was just going to paste that |
| 17:36.34 | raden | I cannot find anything that says SPI or firewall disable or anything of the such in new firmware |
| 17:38.49 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 17:38.58 | [TK]D-Fender | just means they buried it even DEEPER |
| 17:39.32 | raden | yeah im looking :( |
| 17:39.47 | stupidnic | Another point of note, I just tried placing a call Zap to Zap (FXS to FXO) and the call connected as it rang, but there was zero audio on my end (the ringing tone) |
| 17:39.53 | raden | loooked under security looked under NAT looked under QOS |
| 17:39.54 | *** join/#asterisk trebaum (n=trebaum@ip68-8-175-208.sd.sd.cox.net) |
| 17:39.55 | pryorda | thanks guys |
| 17:39.55 | *** part/#asterisk pryorda (n=dpryor@unaffiliated/irated) |
| 17:39.59 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 17:40.14 | drmessano | needs a store discount coupon for the digium Skype app |
| 17:40.19 | bmoraca | it's usually under "WAN" or something like that... |
| 17:40.27 | drmessano | tried SSSSSKYPE and got nuthin |
| 17:40.34 | stupidnic | is it possible that my TDM card is going bad? or am I dealing with a wiring issue? |
| 17:40.40 | bmoraca | however, the reference manual for that firewall only references "SPI" once |
| 17:40.49 | trebaum | I'm having a compiling problem. If I compiled the newest version of asterisk, and now its complaining in the messages log that there is a dahdi config issue, will that stop the asterisk daemon from starting? |
| 17:41.11 | trebaum | i'm trying to figure out if its a config issue, or a compiling issue. |
| 17:41.52 | drmessano | It shouldnt |
| 17:41.52 | raden | drmessano, [TK]D-Fender , is there something besides SIP or SPI or firewall i should be looking for |
| 17:42.20 | drmessano | Asterisk compiles with dahdi *support* when present, it doesnt make it dependency |
| 17:42.25 | bmoraca | a new router? |
| 17:42.36 | [TK]D-Fender | raden: Look everywhere.... or do the smart thing and replace it first, validate that things work without it and the do whatever you want |
| 17:42.40 | *** join/#asterisk Strogg (n=jean@unaffiliated/strogg) |
| 17:42.42 | drmessano | bmoraca: Shoosh, he barely broke this on |
| 17:42.42 | Strogg | 'lo 'lo |
| 17:42.42 | drmessano | bmoraca: Shoosh, he barely broke this one |
| 17:43.10 | *** join/#asterisk Skeeter- (n=wil_c_wi@190-141.cgocable.ca) |
| 17:43.16 | bmoraca | i haven't had good luck with NetGear's "small business" crap |
| 17:43.21 | Strogg | how do you know which module you need to load in your modules.conf, in order to get access to a certain cmd? |
| 17:45.44 | raden | brb throwing the linksys in place |
| 17:45.45 | [TK]D-Fender | Strogg: Most are largely clear. |
| 17:46.00 | trebaum | it was a dahdi config issue. Thanks for the help. |
| 17:46.02 | *** part/#asterisk raden (n=tanning@69-179-99-17.stat.centurytel.net) |
| 17:46.02 | Strogg | let's say it was Playtones() |
| 17:46.17 | drmessano | I've found a good SOHO box can be made into a good next-step-up box with alternate firmware.. but that too has its limits |
| 17:46.24 | Strogg | [TK]D-Fender: I've been browsing the wiki, and it doesn't seem to say which module is needed to load the command |
| 17:46.33 | drmessano | and that the average "next step up" box is total crap |
| 17:46.45 | drmessano | Worse than the $50 model |
| 17:46.53 | drmessano | Goes for Linksys, Netgear, Dlink |
| 17:46.54 | [TK]D-Fender | Strogg: Get grepping |
| 17:47.07 | Strogg | alrighty.. |
| 17:47.19 | [TK]D-Fender | drmessano: No... D-Link starts sucking right from 0$ :) |
| 17:47.56 | drmessano | I stand corrected |
| 17:48.16 | drmessano | Touche`, or should I say, `Douche? |
| 17:48.18 | *** join/#asterisk errotan (n=errotan@62.201.123.198) |
| 17:55.53 | *** join/#asterisk raden_work (n=tanning@69-179-99-17.stat.centurytel.net) |
| 17:57.34 | *** join/#asterisk youngproguru (n=youngpro@74.10.229.45) |
| 17:57.38 | raden_work | [TK]D-Fender, wrt54gl is bridged to DSL modem via wan port all is working back to WIFI phone issue |
| 18:00.25 | raden_work | [TK]D-Fender, drmessano , www.voltarclamps.com/files/sip.txt |
| 18:02.27 | bmoraca | so did asterisk register with callcentric after you swapped out routers? |
| 18:02.50 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 18:03.10 | *** join/#asterisk raden_work (n=tanning@69-179-99-17.stat.centurytel.net) |
| 18:07.15 | raden_work | brb |
| 18:12.18 | lowtek | "One type-o to rule them all!" by LowTek |
| 18:12.27 | lowtek | "How to bring Asterisk to it's knees" by LowTek |
| 18:13.20 | *** join/#asterisk raden_work (n=tanning@69-179-99-17.stat.centurytel.net) |
| 18:17.29 | raden_work | [TK]D-Fender, Transmitting (NAT) to 204.11.192.37:5060: |
| 18:17.29 | *** join/#asterisk lucasb (n=bussey@office.telifon.com) |
| 18:17.30 | raden_work | [TK]D-Fender, www.voltarclamps.com/files/sip.txt |
| 18:17.39 | [TK]D-Fender | raden_work: Check your forwarding. |
| 18:17.56 | raden_work | 5060 - 20000 forwarded |
| 18:19.17 | [TK]D-Fender | raden_work: Something is just flat-out FUBAR'd on your side... |
| 18:19.22 | [TK]D-Fender | raden_work: I'm not sure what ATM |
| 18:19.27 | lowtek | ~fubar |
| 18:19.27 | infobot | fubar is F*cked Up Beyond Any Recognition, e.g. "This whole operation is fubar, soldier" (gay lisp included), or a bar addon like Titan Panel and Telo's InfoBar. and everything. |
| 18:19.43 | raden_work | everything working except that freaking wifi phone it just doesnt seem logical |
| 18:19.55 | raden_work | [TK]D-Fender, could it be the linksys wifi router ? |
| 18:20.03 | raden_work | should i try my buffalo ? |
| 18:20.20 | [TK]D-Fender | raden_work: Is that what you used to replace your netgear? |
| 18:20.41 | raden_work | Linksys WRT54GL w/ DDWRT micro replaced Netgear with new firmware |
| 18:21.07 | [TK]D-Fender | raden_work: Your attempts to be "smart" are failing bad... |
| 18:21.10 | raden_work | and im having the same problem i was before i replaced the netgear firmware and the netgear became useless |
| 18:21.13 | [TK]D-Fender | raden_work: Do something NORMAL FFS |
| 18:21.21 | raden_work | Normal FFS ? |
| 18:21.29 | [TK]D-Fender | ~ffs |
| 18:21.30 | infobot | rumour has it, ffs is for f**k's sake, or for fine's sake. UCB's Fast File System |
| 18:21.43 | raden_work | what did i do that was not normal ? |
| 18:21.52 | raden_work | replaced the netgear with the linksys |
| 18:22.05 | raden_work | the fixed my registration problem |
| 18:22.27 | raden_work | but still have the same wifi issue only remaining variable in the linksys and the phone itself |
| 18:22.30 | lowtek | fast file system? Asterisk runs on the Amiga? |
| 18:22.39 | lowtek | raden_work: STEP ON THE F*CK |
| 18:22.45 | lowtek | raden_work: problem solved |
| 18:22.46 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:22.49 | lowtek | ~sotf |
| 18:22.51 | raden_work | lowlevel, lmao |
| 18:22.58 | guax | look, i have one AGI, that uses the SET EXTENSION command, when i set the extension and the call is answered the CDR stores the right value. when CANCEL happens CDR dont change the callerid. thats a way to force this? |
| 18:23.16 | guax | theres a way* |
| 18:25.03 | [TK]D-Fender | raden_work: You're running cooked Firmware on that thing |
| 18:25.29 | *** join/#asterisk matt_d (n=matt@70.134.79.103) |
| 18:25.37 | matt_d | hello everyone |
| 18:25.44 | trebaum | hola matt |
| 18:26.14 | raden_work | its not the firmware can we continue or do i need to change it back to make the world a better place ? |
| 18:26.14 | matt_d | trebaum: what's going on? |
| 18:27.16 | matt_d | let me run something by everyone. i just woke up, so i havent tried this yet: |
| 18:28.03 | [TK]D-Fender | raden_work: I'mjust saying it like it is. You aren't running advisable hardware, and for the stuff that is, you're running custom stuff on top of it. Backing this with unfaltering faith is a recipe for failure. |
| 18:28.10 | matt_d | chanspy's whisper function is broke. horrible delay and digium refuses to fix it... now, if i bridge two calls togeahter SIP/1000 & SIP/1001 then create a call file to dial out and bridge SIP/1002 to SIP/1001 will all users hear voice from SIP/1000 or will SIP/1001 just hear it? |
| 18:28.24 | raden_work | [TK]D-Fender, give me 5 min ill reload to linksys |
| 18:28.26 | matt_d | im trying a workaround for chanspy's broken whisper function |
| 18:28.27 | [TK]D-Fender | raden_work: You should be simplifying rather than complicating your compounding issues. |
| 18:28.39 | drmessano | [TK]D-Fender: He has a screwed up bridge in the NVRAM of the WRT54GL.. Which is where I was headed for 5 hours last night.. If he performed a proper reset, this would all be over |
| 18:28.52 | raden_work | ddwrt micro to me is simple linksys i have had more issues with than ddwrt but ill switch back |
| 18:28.53 | trebaum | matt_d: another day another voip config issue. :) |
| 18:29.01 | raden_work | and what router should i be using for this stuff ? |
| 18:29.01 | stupidnic | anybody have a 102 milliwatt test number I can test with? |
| 18:29.02 | drmessano | His WIFI is acting NAT'd.. that says it all |
| 18:29.27 | raden_work | drmessano, how does that say it all ? |
| 18:29.40 | trebaum | matt_d: I don't know anything about it. |
| 18:29.42 | drmessano | Switching back to the Linksys firmware wont fix shit without a proper reset |
| 18:29.50 | trebaum | matt_d: i'm still a newb |
| 18:29.54 | raden_work | omg lets not get into this ok |
| 18:29.57 | drmessano | You can change the firmware, but the NVRAM isnt moving |
| 18:30.06 | raden_work | i get where u coming from |
| 18:30.08 | matt_d | trebaum: well i will find out now :) |
| 18:30.28 | raden_work | i have a brand new buffalo in the box im just going to open that up and throw that in linksys place |
| 18:30.45 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 18:31.03 | *** join/#asterisk fnord0 (n=fnord0@unaffiliated/fnord0) |
| 18:32.39 | cusco_ | hi |
| 18:32.46 | matt_d | hi cusco_ |
| 18:32.59 | cusco_ | can't figure what is wrong with pri: http://paste.debian.net/44519/ |
| 18:33.23 | cusco_ | that happens when I try to outbound |
| 18:33.27 | bmoraca | god i wish i could find someone who's actually used a Lucent MAX TNT as a pri-to-sip gateway before...there is precisely 0 information about it on the interwebs |
| 18:33.28 | cusco_ | it goes unreacheble |
| 18:34.38 | [TK]D-Fender | bmoraca: I know a few people who've passed through here have touched one before. I *think* one was AsteriskMonkey |
| 18:36.05 | carrar | MAX TNT make great heaters |
| 18:36.09 | bmoraca | we've used them for a number of years for our dialup ISP...but we outsource that now, so we were going to take our current units and put them to use in this respect...they're ridiculously obscure, though, and i can't find any docs for them later than 10.0, but you don't get sip support until 10.0.2 or 10.1 (conflicting reports) |
| 18:36.19 | carrar | space heater / coffee table |
| 18:36.34 | bmoraca | whitenoise generator |
| 18:37.21 | *** join/#asterisk ZaVoid (n=zavoid@75-147-121-177-Philadelphia.hfc.comcastbusiness.net) |
| 18:37.27 | ZaVoid | hi all |
| 18:37.48 | cusco_ | hi |
| 18:37.49 | bmoraca | i just figure that using one of these will be a bit safer than using an Asterisk box with 24 PRI ports in it...at least as far as stability goes... |
| 18:38.32 | lowtek | bmoraca: We have a stack of asterisk servers with digium pri cards, 96 total pri's, works great. |
| 18:39.45 | ZaVoid | so if i got a UA that supports g.723 only.. and a endpoint/peer that supports g.729 and g.723...(and in sip.conf allowed for g723/g729). asterisk sends both codec's to the endpoint even though the originating UA doesn't support g729 and asterisk can't transcode g723/g729 |
| 18:40.12 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:40.37 | bmoraca | how many PRIs per server? our goal is to act as a PSTN gateway for our hosted PBXes... |
| 18:41.17 | lowtek | bmoraca: That's basically what we do with fail-over via sip to alternate gateways... lemme ask, I'm not on the hardware side, one min... |
| 18:42.49 | [TK]D-Fender | ZaVoid: Because you told it to allow both. So DON'T |
| 18:43.05 | ZaVoid | Hi Fender |
| 18:43.26 | ZaVoid | of course theres more to it, for example the far end carrier accepts both.. and some UA's will support g729 and some g723 and only send one or the other |
| 18:44.34 | ZaVoid | i found this. http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch anyone ever use it? |
| 18:44.42 | lowtek | bmoraca: 6 servers with 4 TE4120 cards and 1 TCE400B card in each |
| 18:45.26 | *** join/#asterisk drclue (n=drclue@ip-65-49-160-6.wireless.dyn.beamspeed.net) |
| 18:45.27 | drmessano | TCE400B? Drool |
| 18:45.31 | bmoraca | lowtek: how do you handle the failover? application-level or are you doing something like DUNDI and manually configuring the failover peers in the dialplan? |
| 18:45.39 | *** join/#asterisk af_ (n=getsmart@88-149-240-216.dynamic.ngi.it) |
| 18:46.10 | lowtek | bmoraca: I'm not sure on those boxes, we do use DUNDI internally, I think DUNDI is the greatest thing since asterisk but it never gets the credit it deserves or the docs |
| 18:46.12 | bmoraca | and how do the TCE400Bs work? i'm looking at potentially getting the PCI version |
| 18:47.02 | lowtek | bmoraca: It works, not sure about the specifics, I guess you install a module for it for g729 transcoding. Those guys are in Dallas, TX and I'm in Birmingham, AL. |
| 18:47.15 | bmoraca | ahh |
| 18:47.40 | lowtek | lowtek: I do know that we support g729 without any software transcoding througout ... |
| 18:48.02 | bmoraca | right now, i don't have a need for the redundancy of multiple gateways, but i'm quickly approaching it, and i'm researching different ways to do it |
| 18:48.05 | lowtek | s/lowtek/bmoraca <- talking to myself |
| 18:48.21 | bmoraca | yeah, that's what the TCE400B is for...hardware transcoding and not having to worry about licensing |
| 18:49.53 | drclue | Howdy All. I'm looking for some guinea pigs for a new lightweight FastAGI/AMI daemon bridge to PHP. The the tool dynamically loads PHP scripts at dial time based upon the dialplan and provides interfaces to both AGI commands and AMI commands. Additional features will be provided based upon interest. Any takers? |
| 18:51.14 | bmoraca | i don't want to have to worry about maintaining multiple trunk groups between my gateways, and it'd be nice to be able to use a single IP to reference all of the gateways. I've been looking about layer 7 load balancing, but I don't know how well that'll work, as i don't believe it's possible to share SIP registries between multiple asterisk boxes and that'd be required for a layer 7 load... |
| 18:51.16 | bmoraca | ...balancing appliance to work |
| 18:52.53 | drclue | FastAGI could help you load balance |
| 18:53.54 | matt_d | drclue: hey there! |
| 18:54.17 | drclue | Howdy Matt_d , glad to see ya got chanspy happy |
| 18:54.37 | matt_d | drclue: so i thought.. it *was* working but now its back to its old ways |
| 18:54.53 | drclue | Dam sucker punched again |
| 18:54.57 | matt_d | drclue: now im trying to study app sources to possibly make my own .. |
| 18:55.20 | lowtek | matt_d: ChanSpy() on 1.4 or 1.6? |
| 18:55.35 | matt_d | lowtek: 1.6 have also tried it on 1.4 |
| 18:55.46 | matt_d | lowtek: the function works, but the whisper feature is delayed .. |
| 18:55.50 | matt_d | which is really bad |
| 18:56.02 | lowtek | matt_d: It's broke on all of my 1.4.26 installs, I had to noload it, it core-dumps asterisk. |
| 18:56.04 | matt_d | can be real time, 5 seconds, 10 or 15 seconds. really annoying |
| 18:56.58 | *** join/#asterisk nightrid3r (n=kvirc@78-20-232-172.access.telenet.be) |
| 18:57.00 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 18:57.41 | WHYS | can someone confirm - Dial(SIP/ch1/22&SIP/ch2/33) will connect the first answered call, but only *after* both calls are connected. (if one channel is not functioning, the other channel is never connected) |
| 18:58.18 | drclue | While I've not yet tried it ,bu as a programmer I like this idea as it seems to get the recording going ahead of the spy which *might* be of value. |
| 18:58.18 | drclue | exten => _*29XXXX,1,Answer |
| 18:58.18 | drclue | exten => _*29XXXX,n,set(calltime=${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)}) |
| 18:58.18 | drclue | exten => _*29XXXX,n,MixMonitor(/var/spool/asterisk/monitor/X${calltime}X${CALLERID(num)}X${EXTEN:3}X.wav) |
| 18:58.18 | drclue | exten => _*29XXXX,n,Chanspy(SIP/${EXTEN:3}|q) |
| 18:58.28 | [TK]D-Fender | drclue: PASTEBIN, do not spam |
| 18:59.18 | drclue | Ya ya , pastebin. four lines seemed more conversational than pastebin and certainly not spam |
| 18:59.33 | drclue | but I'm sure you'll argue the point |
| 19:00.28 | lowtek | drclue: Best store as ${UNIQEID}.wav and update ${USERFIELD} with the ${UNIQUEID} otherwise you'll have a bitch of a time finding them later without some serious parsing of dirs |
| 19:01.25 | drclue | lowtek, your probably correct. It's just I had promised matt_d I would look into this and after hours of searching this seemed the most interesting snippet |
| 19:01.51 | *** join/#asterisk sezuan (i=sezuan@mobil5.vp.ip6.scheff32.de) |
| 19:02.05 | matt_d | drclue: the recording works, still falls down to the delayed 'whisper' function |
| 19:02.54 | matt_d | is so frustrated with ChanSpy/ExtenSpy that he would switch over to CallWeaver if their ChanSpy/ExtenSpy works .... |
| 19:03.09 | drclue | matt_d , I was sorta hoping that getting the recording happening ahead of the chanspy might be useful , but then again , maybe forking it off would work even better. |
| 19:03.26 | matt_d | drclue: i havent tried forking yet .. |
| 19:03.50 | lowtek | matt_d: I would always StopMonitor() before ChanSpy() because that just seemed like a lot happening at the same time. |
| 19:04.12 | matt_d | even if i don't record, it still has a delay. |
| 19:05.32 | drclue | Your *always* going to have *some* delay , it's just trying to get that delay to be reasonable |
| 19:05.54 | matt_d | the delay is up to 15 seconds :) |
| 19:06.06 | drclue | I know , the 15 seconds sucks |
| 19:06.08 | matt_d | i don't mind a second, but 15 seconds is not acceptable for the project. |
| 19:06.09 | matt_d | hehe |
| 19:06.42 | drclue | Best I can figure is to get the recording going ahead of the spy |
| 19:06.51 | lowtek | Wow! Amazing how much better asterisk runs when you don't have thousands of unclosed MySQL connection pointers stacked up. |
| 19:07.12 | drclue | That seemed to be the jist of the SPAM FOUR LINES I pasted above :) |
| 19:07.35 | *** join/#asterisk raden_work (n=tanning@69-179-99-17.stat.centurytel.net) |
| 19:07.46 | raden_work | ok tried buffalo same thing |
| 19:07.48 | matt_d | drclue: yesterday i had a test run of about 45 minutes. worked great, except the delay. recording went swell. |
| 19:08.16 | [TK]D-Fender | raden_work: Same thing, what? that * still bombs with your ITSP? |
| 19:08.19 | lowtek | matt_d: You could have a seperate MixMonitor() and ChanSpy() servers, that would do it. |
| 19:08.33 | raden_work | reflashed Linksys to newest firmware, reset factory defaults, unplugged for 2 minutes plugged back in setup PPOE, Disabled firewall, setup port forwarding still same thing |
| 19:08.46 | drclue | matt_d , I'm pretty positive if one can get the recording going ahead of the spy via a separate thread , life will be good |
| 19:08.50 | raden_work | [TK]D-Fender, one way audio on WIFI phone |
| 19:09.04 | raden_work | [TK]D-Fender, netgear is the issue with ITSP |
| 19:09.06 | matt_d | lowtek: i tried chanspy on a solo machine, still a delay. seems like there is something within asterisk that is delaying it. its not bandwidth or processing power. i even tried to mak all connections ulaw (so there is no converting) and still . |
| 19:09.07 | [TK]D-Fender | raden_work: are your OTHER problems solved? |
| 19:09.16 | drclue | WIFI = NAT , all my phones are WiFi |
| 19:09.25 | [TK]D-Fender | raden_work: Because we got the WIFI working early this morning. |
| 19:09.32 | lowtek | matt_d: What about with ChanSpy(XX|bq)? |
| 19:09.43 | raden_work | [TK]D-Fender, it works in office as in phones on the lan |
| 19:09.59 | raden_work | if i call out i cant here anyone |
| 19:10.07 | raden_work | if someone calls in i have audio both ways |
| 19:10.10 | matt_d | lowtek: havent tried the 'b' option yet. i will try |
| 19:10.16 | lowtek | matt_d: So it only looks for bridged calls, I think it uses audiohooks that way, not sure. |
| 19:10.40 | [TK]D-Fender | raden_work: And can your fixed phones call out and get audio? |
| 19:10.56 | raden_work | yes sir |
| 19:11.11 | *** join/#asterisk flujan (n=flujan@189.111.254.251) |
| 19:11.11 | raden_work | all wired phones work great been using all this week without issue |
| 19:11.20 | drmessano | Youre not doing a proper reset if you think unplugging for 2 mins does shit |
| 19:11.35 | matt_d | here is what its doing. user logs in and enteres telephone #1 and telephone #2. user is placed into chanspy and a call file is created to call telephone #1 and telephone #2. the call file has SPYID set to a unique id so only the user will be in the chan spy. |
| 19:12.12 | raden_work | drmessano, what do i have todo |
| 19:12.19 | drmessano | For the last time |
| 19:12.24 | drmessano | Hard Reset (aka 30/30/30 reset): |
| 19:12.24 | drmessano | The following procedure will clear out the NVRAM and set dd-wrt back to default values: |
| 19:12.24 | drmessano | With the unit powered on, press and hold the reset button on back of unit for 30 seconds |
| 19:12.24 | drmessano | Without releasing the reset button, unplug the unit and hold reset for another 30 seconds |
| 19:12.25 | drmessano | Plug the unit back in STILL holding the reset button a final 30 seconds (please note that this step can put Asus devices into recovery mode...see note below!) |
| 19:12.26 | drmessano | This procedure should be done BEFORE and AFTER every firmware upgrade/downgrade. |
| 19:12.43 | drmessano | **STILL HOLDING** |
| 19:13.39 | *** join/#asterisk Defraz (n=T0tal@corp.fuzecore.com) |
| 19:13.51 | *** join/#asterisk semaries (n=martin@stgt-5d84918a.pool.einsundeins.de) |
| 19:14.01 | matt_d | would b option work? is telephone #1 and #2 bridged? |
| 19:15.46 | raden_work | drmessano, give me a few minutes trying to get wifi phone firmware updated |
| 19:15.59 | raden_work | the comapny called me back admiting they have issues ;( |
| 19:18.01 | drmessano | Collect call from China? |
| 19:18.09 | drmessano | Thats gonna hurt the project |
| 19:18.28 | *** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker) |
| 19:18.34 | drmessano | Asterisk box made from recycled workstation: $10 |
| 19:18.48 | drmessano | Router we had in the closet at my moms: $0 |
| 19:18.57 | drmessano | Phones we got used off ebay: $200 |
| 19:19.16 | matt_d | ... please not an AmEx commercial |
| 19:19.22 | *** join/#asterisk Laibsch (n=Laibsch@p5B3B3E25.dip.t-dialin.net) |
| 19:19.23 | drmessano | Cost of collect call from China to support the $30 wifi phone my boss HAD to have: $2750 |
| 19:19.30 | drmessano | The LULZ: Priceless |
| 19:19.51 | lowtek | lol |
| 19:19.53 | TSM2 | ouch |
| 19:20.11 | TSM2 | :) |
| 19:20.26 | Naikrovek | i've never heard a good comment about a wifi voip phone |
| 19:20.27 | drmessano | I just summed up raden_work's week |
| 19:20.44 | *** join/#asterisk sjobeck (n=Adium@32.153.125.37) |
| 19:20.59 | *** part/#asterisk sjobeck (n=Adium@32.153.125.37) |
| 19:21.05 | Naikrovek | it's F'd up that you think you can sum up someone else's work week that easily |
| 19:21.43 | raden_work | drmessano, thanks for the ripping on as usual |
| 19:21.43 | carrar | heheh |
| 19:22.02 | drmessano | He spent all of it in here.. It was like an emo twitter/live journal mashup |
| 19:22.11 | Nugget | hah |
| 19:22.55 | raden_work | yeah nice |
| 19:23.04 | J4k3 | Naikrovek: its the phones. I've had much better luck with good wireless bridges than I have a 'wifi handset' of any sort. |
| 19:23.18 | Naikrovek | yeah that's what i figured. |
| 19:23.30 | raden_work | anyone recomend a good router ? with 8 ports or more |
| 19:23.38 | Laibsch | Hi, I hope you don't mind a question from a non-technical person. I'm a freelancer that travels around quite a bit. I'd like to be able to use VoIP to call out from my home telephone network. (computer - voip - asterisk - home PSTN, that's the idea). |
| 19:23.44 | J4k3 | raden_work: a router is a router, a switch is a switch. ;) |
| 19:23.45 | Naikrovek | my Indian office wants four wireless phones, and they scoffed when I told them to get traditional phones with an ATA |
| 19:24.00 | drmessano | "This phone sucks. I am going to reset the Linksys, screw up my config, and reboot. #zomg #router #chinese_wifi_phone" |
| 19:24.00 | Laibsch | I need both telephone and fax, single line is sufficient. Either analog or ISDN. |
| 19:24.03 | J4k3 | Naikrovek: so stick an ATA and some DECT bases in a 'box' for them |
| 19:24.05 | drmessano | Was that 140? |
| 19:24.09 | J4k3 | and stick a HUGE price tag on it |
| 19:24.29 | Naikrovek | J4k3: they can't use phones over 3ghz in India for some reason, but I told them to buy the same locally |
| 19:24.36 | Laibsch | Will a card such as the X100p on ebay be up to that task or do I need something else? |
| 19:24.43 | raden_work | drmessano, whatever |
| 19:24.48 | Laibsch | Is this even something recommendable to pursue? |
| 19:25.09 | lowtek | Laibsch: TDM400p at least |
| 19:25.12 | Nugget | Laibsch: x100p is nothing more than a "proof of concept" you can play around with before deciding to buy something that doesn't totally suck. |
| 19:25.19 | J4k3 | Naikrovek: DECT is 1.9ghz, but the spectrum license doesn't exist in a lot of markets (and varies from market to market. IE - EU DECT is a completely different frequency than US DECT |
| 19:25.51 | Naikrovek | ah |
| 19:25.51 | drmessano | Testing asterisk with an x100p is like testing a laptop with Vista. Make sure it works, then load something real on it |
| 19:25.54 | J4k3 | for a single line you're usually better off getting an ATA with an FXO port. |
| 19:26.04 | J4k3 | TDM400's ain't free/cheap |
| 19:26.05 | Naikrovek | well I told them to buy phones locally because shipping doubles the cost anyway |
| 19:26.09 | Nugget | J4k3 is correct. |
| 19:26.26 | Nugget | an ATA also dodges having to futz with drivers |
| 19:26.30 | J4k3 | yep |
| 19:26.40 | J4k3 | "oh no my VIA chipset mobo wants to eat half my TDM data!" |
| 19:27.01 | drmessano | Aint nothing wrong with a VIA chips.... ok, yeah, there is |
| 19:27.11 | J4k3 | hehe |
| 19:27.15 | lowtek | I have two TDM400P's with FSO cards, for sale, cheap, $150 takes both. |
| 19:27.24 | Laibsch | lowtek, Nugget: so, I'll be looking at several hundred $ of hardware at the minimum? What is the x100p missing? Again, it's just me using this. one person, one line. |
| 19:27.31 | drmessano | VIA 4-in-1 drivers = 4 ways to screw your system with 1 easy installer |
| 19:27.43 | Naikrovek | are you the same Nugget that used to hang out in #slashdot? |
| 19:27.43 | J4k3 | Laibsch: a FXO capable ATA is only about $60 USD |
| 19:27.51 | J4k3 | and will come with an FXS port too. |
| 19:27.52 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 19:27.56 | Laibsch | nice |
| 19:28.06 | J4k3 | a good one might cost a little more, but you're looking at under $100 |
| 19:28.11 | KyleK | yay SPA3102 |
| 19:28.18 | drmessano | SPA-3102 |
| 19:28.18 | Laibsch | J4k3: any suggestions |
| 19:28.19 | Laibsch | ? |
| 19:28.21 | J4k3 | and no need to deal with a PCI card/drivers/etc |
| 19:28.23 | Nugget | Laibsch: the sooner you stop viewing asterisk as a low-cost solution and instead start viewing asterisk as a flexible solution the better off you'll be. |
| 19:28.33 | Nugget | like all things in the world, you do tend to get what you pay for |
| 19:29.07 | Laibsch | Nugget: Well, but I kind of want to know what I'm buying. And why I need it. |
| 19:29.15 | Laibsch | I trust your judgment |
| 19:29.22 | Laibsch | But I'd like to understand myself, too |
| 19:30.02 | drmessano | Just like putting Linux on a real server to create a powerful solution, not throwing your free OS on a POS machine in the closet to use for the company fileserver, web server, HR database server, etc.. |
| 19:30.09 | bmoraca | i have a hosted pbx customer using a couple of spa3102s to connect some leased lines as trunks to their hosted pbx...work pretty well |
| 19:31.04 | TSM2 | look at the SPA8000 now, all depends on how many lines you want |
| 19:31.23 | bmoraca | SPA8000 is FXS only |
| 19:31.25 | drmessano | "I downloaded Ubuntu, almost ran over this PC driving home drunk on Monday night, and whaddyaknow, I got a file server now" |
| 19:31.26 | KyleK | is there a sip ping command available? I'm wondering if my port 5060 is blocked |
| 19:31.35 | drmessano | FOSS FAIL |
| 19:32.03 | *** join/#asterisk Tim_Toady (n=moi@adsl52-231.kln.forthnet.gr) |
| 19:32.38 | KyleK | installing linux onto a vespa is the same commands and configuration stuffs as installing linux on a tank |
| 19:32.48 | bmoraca | KyleK: are you connecting TO SOMETHING or is something connecting TO YOU? |
| 19:32.55 | Naikrovek | does the tank have internet? |
| 19:33.04 | KyleK | bmoraca: i want something to connect to me |
| 19:33.11 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 19:33.15 | KyleK | the road is the internet |
| 19:33.49 | bmoraca | did you configure your firewall to open the port? |
| 19:33.55 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:33.58 | drmessano | KyleK: Yeah, but taking home a free tank to make your home file server just because it was offered to you, is in a word, ghettp |
| 19:34.06 | drmessano | ghetto* |
| 19:34.08 | Naikrovek | what if the road was the internet. some 2D barcode on the surface of the road that had data on it that your vehicle could read as it drove over it. |
| 19:34.26 | Naikrovek | containing construction information or local tourism info or something |
| 19:34.38 | Naikrovek | impractical maybe |
| 19:34.47 | bmoraca | Naikrovek: why would you need that when GPS can do it much more simply? |
| 19:35.08 | drmessano | I threw out a bunch of 1GHZ boxes I had laying here.. Perfectly usable. Why? I like the color of my carpet |
| 19:35.19 | Naikrovek | GPS gives you location, not trivia about the town you're in. I guess this may have been a good idea if the internet and wifi never came around |
| 19:35.23 | KyleK | bmoraca: well i can access it sometimes remotely |
| 19:36.13 | bmoraca | Naikrovek: your GPS receiver can use that location information to do anything...including tell you about the location on which you're standing |
| 19:36.45 | Naikrovek | i still think a wifi mesh of cars would be best |
| 19:36.57 | *** join/#asterisk DrkShadow (n=andrew@host-72-175-240-62.static.bresnan.net) |
| 19:37.02 | Naikrovek | bmoraca: i got it - it's not a great idea. |
| 19:37.07 | Naikrovek | bmoraca: i agree |
| 19:37.49 | *** join/#asterisk lowtek (n=anonymou@mail.heavylogic.com) |
| 19:38.04 | J4k3 | drmessano: I've found myself selecting stuff to junk based on its performance/watt ratio |
| 19:38.07 | bmoraca | woo...i get to build a Core i7-based HP ML350 G6 tomorrow! first one. it'll be fun. |
| 19:38.16 | *** join/#asterisk davidandgoliath (n=David@S0106001d60d4e488.vn.shawcable.net) |
| 19:38.16 | Naikrovek | nice |
| 19:38.18 | DrkShadow | Hey, I'm trying to replace a phone. The phone I'm replacing is Aastra 57iCT/2.0.1.1076 Brcm-Callctrl/v1.7.2.2 MxSF/v3.6.2.5, and I'm replacing it with Aastra 57i/2.5.0.82. I did a factory defaults on the new phone, and then I set it up exactly like the previous. When I try to make a call, it _always_ fails, but can receive calls fine. Any ideas? |
| 19:38.40 | Naikrovek | wishes he got a Core i7 BladeCenter server instead of the c2q he got |
| 19:39.01 | J4k3 | i7's are nice |
| 19:39.11 | Naikrovek | even with c2q I've got 10 virtual machines on that one physical machine |
| 19:39.39 | J4k3 | now if the i7 just had reasonably priced motherboards available... |
| 19:39.55 | Naikrovek | yeah |
| 19:39.55 | bmoraca | i'm building this as an exchange server. a bit overpowered, but this client doesn't replace servers often, so i figured overpowered would fit the bill, because it'll be just right in 10 years when it's falling apart and needs to be replaced again. |
| 19:40.19 | Naikrovek | my exchange server lives on a virtual machine - runs great. i like ex2007 btw |
| 19:40.24 | Naikrovek | but that's offtopic |
| 19:40.28 | J4k3 | yeah. I never understand those kinds of companies |
| 19:40.40 | Naikrovek | i understand them: bottom line rules |
| 19:40.44 | bmoraca | i'm ticked at HP for not having a Core i7-based ML310. i can't sell servers for $2500 anymore. |
| 19:40.51 | J4k3 | its cheaper to replace 'reasonable' every 3 years than buying a megaserver (that will likely require service during its lifetime) every 10 years |
| 19:41.00 | bmoraca | Naikrovek: not when they're paying me $250 every other week to fix some issue on it |
| 19:41.06 | *** join/#asterisk lowtek (n=anonymou@mail.heavylogic.com) |
| 19:41.12 | Naikrovek | yes but long-term isn't something that bottom-line companies can see very clearly |
| 19:41.38 | Naikrovek | what was bleeding edge 10 years ago, what is this server replacing |
| 19:41.41 | Naikrovek | exchange 5.5 probably |
| 19:41.45 | Naikrovek | maybe 4.0 |
| 19:41.46 | bmoraca | i had one customer that must have paid us $3000 over the last year to fix his laptop that he kept breaking instead of just buying a new one. |
| 19:41.53 | Naikrovek | single p3 1ghz |
| 19:42.00 | drmessano | J4k3: I agree completely.. Too much changes to buy anything that far out, and the assumption that buying better-than-top-of-the-line now for longevity is nonsensical |
| 19:42.08 | bmoraca | nah, we upgraded it to exchange 2000 a while ago...it's a ML350 G1 |
| 19:42.20 | lowtek | Exchange is the only ms server box we have company-wide, everything else is linux. |
| 19:42.52 | bmoraca | asterisk is the only thing we run on linux |
| 19:42.52 | Naikrovek | exchange is really nice these days, as is windows server 2008 |
| 19:43.00 | drmessano | Moving away from Exchange is the best thing we ever did |
| 19:43.08 | bpgoldsb | Will 'extensions reload' back-out safely if it detects an error? |
| 19:43.10 | bmoraca | er...not entirely true...my DNS blackhole servers are also linux |
| 19:43.11 | tomodachi | drmessano: what did you move to? |
| 19:43.13 | Naikrovek | drmessano: what did you move to |
| 19:43.16 | Qwell | drmessano: psst. test a yum update for me. |
| 19:43.19 | bpgoldsb | Or is there a way to pre-check your changes? |
| 19:43.42 | drmessano | Combination of hosted services.. Google Apps primarily.. Not perfect, but far less of a nightmare |
| 19:43.53 | Naikrovek | well licensing is cheaper that way |
| 19:43.55 | tomodachi | ok |
| 19:43.59 | Naikrovek | Exchange CALs are murder |
| 19:44.06 | drmessano | Ex2007 is a beast, and its VERY VERY beta quality |
| 19:44.16 | Naikrovek | pfft |
| 19:44.18 | Naikrovek | works great for me |
| 19:44.20 | Naikrovek | no probelms |
| 19:44.23 | tomodachi | arent the exchange cals a onetime fee? |
| 19:44.25 | bmoraca | i like google apps. it's not a 100% replacement for Exchange/SharePoint, but it's an interesting alternative |
| 19:44.39 | bmoraca | tomodachi: yes, but they're still expensive |
| 19:44.46 | Naikrovek | yes they're one-time, but I paid FAR more in licenses than I did on server hardware and exchange server license |
| 19:44.58 | lowtek | ActiveSync + push email is why we haven't switched to anything else |
| 19:45.10 | tomodachi | lowtek: zimbra does that |
| 19:45.12 | drmessano | The whole certificate model is shit, it requires too much tweaking to keep the Outlook clients from bitching, and even then, theres still major behavioral bugs |
| 19:45.25 | Naikrovek | $19k on Exchange CALs, $5k on hardware and exchange server license |
| 19:45.25 | bmoraca | lowtek: ipswitch's imail supports activesync push, and i've heard it's on the slate for gmail as well |
| 19:46.05 | Naikrovek | drmessano: really? your AD Domain name must have not matched your DNS domain name. that causes all kinds of problems |
| 19:46.18 | drmessano | Want to see a fun one? Set up unified messaging, and then go to the voicemail tab in Outlooks properties.. Prompted to login? Of course.. |
| 19:46.21 | bmoraca | RPC over HTTPS is a bitch to set up under any circumstances. |
| 19:46.29 | drmessano | Naikrovek: Nope, not the issue |
| 19:46.32 | Naikrovek | drmessano: i don't use UM so I can't speak for that |
| 19:47.01 | Naikrovek | drmessano: well I have my outlook clients quieted down, but it took an AD cert server to make it happen |
| 19:47.06 | bmoraca | my UM is having asterisk email voicemails to me, hah! |
| 19:47.20 | drmessano | Naikrovek: How well is that working out for ya with clients using RPC over HTTPS? |
| 19:47.26 | Naikrovek | my "inferior" trixbox install emails my voicemails to me, as well |
| 19:47.52 | bmoraca | Naikrovek: it's not hard to set up in vanilla asterisk... |
| 19:48.21 | Naikrovek | drmessano: works fine, just had to install the root certificate of my ADCS server to the trusted roots on client machines. viola! no more errors |
| 19:48.39 | Naikrovek | now all certs issued by that server are auto-trusted on the domain |
| 19:48.48 | Naikrovek | cured a lot of headaches |
| 19:48.53 | *** join/#asterisk meesterarend (n=frans@vc-41-192-83-127.umts.vodacom.co.za) |
| 19:48.56 | Naikrovek | except for the firefox people (like myself) |
| 19:48.56 | matt_d | in the CLI is there a way to check what codec my iax2 trunk is using? |
| 19:49.24 | drmessano | and I suppose all your home OWA users, you did the same? |
| 19:49.24 | Naikrovek | matt_d: ooh i think so but I can't remember how |
| 19:49.33 | bmoraca | iax2 show peer |
| 19:49.47 | Qwell | drmessano: don't make me get the hose. |
| 19:49.58 | Naikrovek | drmessano: they lived with the error and didnt' complain. they were given the option to install the certificate but no one took it |
| 19:49.59 | drmessano | ? |
| 19:50.06 | drmessano | lol |
| 19:50.09 | Qwell | drmessano: test a yum update for me :p |
| 19:50.10 | Naikrovek | we're too off-topic i guess |
| 19:50.13 | Naikrovek | ah hehe |
| 19:50.20 | matt_d | show peers doesn't show the codec though .. |
| 19:50.20 | drmessano | Which yum update? |
| 19:50.31 | Qwell | on an AsteriskNOW box |
| 19:50.38 | bmoraca | i didn't say SHOW PEERS. i said SHOW PEER |
| 19:50.41 | drmessano | Does that require me to have one? |
| 19:50.46 | Qwell | umm, yes. |
| 19:50.52 | drfreeze | Anyone seen this error: [ResFinderC]: Download - Failed to download file SoundPointIPWelcome.wav, errno 0x380003. |
| 19:50.53 | Qwell | grabs a spork |
| 19:50.59 | Qwell | You DO have one...right? |
| 19:51.00 | Naikrovek | drfreeze: yes |
| 19:51.07 | matt_d | oh.. peer :) hehe |
| 19:51.09 | Qwell | drmessano: also, another doctor. |
| 19:51.14 | drfreeze | Everything seems to be working, but the 550 can't pull down the wav file |
| 19:51.36 | Qwell | drmessano: and he seems much cooler than you. ba-dum-ching |
| 19:51.38 | drmessano | Check my dumpster.. I'm in cleanout mode.. |
| 19:51.41 | drfreeze | Naikrovek: what was the problem when you had that error? |
| 19:51.49 | bmoraca | does that wave file exist in your tftp directory? |
| 19:51.52 | lowtek | So Zimbra is worth looking at? |
| 19:51.59 | Naikrovek | drfreeze: my problem was that the file didn't exist in the directory |
| 19:52.11 | Naikrovek | drfreeze: but putting it there didn't make the phone play the sound |
| 19:52.41 | Naikrovek | check the (t)ftp log to see in what directory it's looking for the file |
| 19:52.44 | Naikrovek | then put the file there |
| 19:52.48 | flujan | someone using iaxmodem and hylafax? |
| 19:52.52 | Naikrovek | if you've done THAT, I dunno |
| 19:53.13 | Naikrovek | if you dont' ahve the file, download the latest firmware, it's in those .zip files |
| 19:53.18 | Naikrovek | knows polycom |
| 19:53.52 | tomodachi | Naikrovek: is it possible to get asterisk working with the vsx7000? |
| 19:54.02 | tomodachi | god knows i´ ve tryed |
| 19:54.05 | bmoraca | flujan: do you have a specific question or are you conducting a poll? |
| 19:54.08 | Naikrovek | is that the conference phone? the ip7000? |
| 19:54.17 | tomodachi | Naikrovek: video conf |
| 19:54.23 | lowtek | flujan: everybody has tried, it's flakey at best, don't expect greater than 90% reliability on fax rc |
| 19:54.26 | lowtek | s/rc/rx |
| 19:54.35 | Naikrovek | never used a video conf phone; what's the issue you're seeing |
| 19:54.51 | tomodachi | i cant get it to work with phonecalls using asterisk at all |
| 19:55.02 | tomodachi | eventhough it supports both sip and h323 (or so it claims) |
| 19:55.04 | bmoraca | lowtek: i haven't noticed any issues receiving or sending faxes, other than that they're slow. and that includes SIP trunking in the middle. |
| 19:55.07 | Naikrovek | but you can get other polycom phones to work? |
| 19:55.30 | lowtek | flujan: There's the one, ask him (bmoraca) |
| 19:55.37 | bmoraca | lol |
| 19:57.24 | Naikrovek | tomodachi: can you get other polycom phones to work? |
| 19:57.37 | Naikrovek | it may be a configuration issue if you're not familiar with polycom phones |
| 19:57.40 | tomodachi | we dont have any polycom phones actually |
| 19:57.48 | flujan | bmoraca: both. I am trying to use it but... well it is not working...Failure to train remote modem at 2400 bps or minimum speed |
| 19:57.51 | tomodachi | just 2 vsx:es |
| 19:58.04 | tomodachi | we hade them before i implemented the asterisk |
| 19:58.04 | Naikrovek | tomodachi: and you're having a problem with just one of them? |
| 19:58.12 | tomodachi | the other one i havent tryed |
| 19:58.32 | flujan | the digium solution only works with digium boards... and I use sangoma on my boxes. |
| 19:58.33 | bmoraca | flujan: what kind of trunking are you using? what codecs? |
| 19:58.52 | tomodachi | Naikrovek: thnx for trying out to help me though, thought there might be a know issue or fw problem or something , i cant give you the exact error since it was a while aago i tryed, if you hang around here maybe i can ask you again later |
| 19:58.55 | flujan | bmoraca: using sangoma E1 board. Alaw codec |
| 19:59.26 | Naikrovek | tomodachi: whisper me your email and I'll send you a quick list of things to check |
| 20:00.23 | tomodachi | done thnx |
| 20:00.53 | flujan | bmoraca: the transmission log if you wish to read http://pastie.org/587625 |
| 20:00.54 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) |
| 20:04.34 | leifmadsen | anyone have an asterisk server I can register against (SIP) for about 10 minutes to test something? |
| 20:04.50 | leifmadsen | I don't have to place any calls -- just attempt to register (it's for issue 15008) |
| 20:05.05 | Naikrovek | got nothing exposed to the 'net i'm afraid |
| 20:05.11 | leifmadsen | np |
| 20:05.24 | leifmadsen | I'd do it on my server, except it is running fsck right now... |
| 20:05.31 | Naikrovek | isn't that nice |
| 20:05.43 | leifmadsen | ya... |
| 20:05.47 | Naikrovek | reboot and it takes 60 minutes because it decides do fsck your 2tb disk that one time |
| 20:06.11 | leifmadsen | "Error reading block 8290019 (Attempt to read block from filesystem resulting in abort read) while doing inode scan. |
| 20:06.24 | Naikrovek | it finds errors though |
| 20:06.27 | bmoraca | flujan: have you tried with plain fax modem and analog line? i'd suspect the issue is more related to hylafax than asterisk |
| 20:06.28 | Naikrovek | which is good |
| 20:06.34 | InfoNutz | hello all again, i was able to catch the '*' when voicemail prompt is playing and send directly to voicemailmain() with the extension but it seems to only pass'a' as the argument. can someone help out, i'm not sure where the value Voicemail("sipcrap@asslkdf","u3334445555") changes to voicemailmain("sip/192910982","a") |
| 20:06.58 | *** join/#asterisk andres833 (n=andres83@190.144.102.122) |
| 20:08.23 | bmoraca | InfoNutz: pastebin what you have and the call log from the CLI |
| 20:08.31 | InfoNutz | k |
| 20:08.42 | drclue | matt_d: for the benefit of others , the issue is likely transcoding that sends your chanspy off into orbit |
| 20:11.11 | InfoNutz | bmoraca: http://www.pastebin.org/10046 |
| 20:11.54 | *** join/#asterisk benneton (n=DELL@adsl-34-157.teol.net) |
| 20:12.03 | benneton | hi guys girls! |
| 20:12.09 | InfoNutz | hello |
| 20:12.19 | bmoraca | InfoNutz: the way you've scripted that, you're getting exactly what you should be |
| 20:12.35 | *** join/#asterisk mnt_real (n=sinan@bas12-montrealak-1167974421.dsl.bell.ca) |
| 20:12.42 | benneton | bmoraca, sa balkana? |
| 20:12.53 | bmoraca | no |
| 20:12.59 | benneton | oket |
| 20:13.00 | benneton | :D |
| 20:13.00 | bmoraca | i guess |
| 20:13.02 | bmoraca | maybe? |
| 20:14.02 | bmoraca | InfoNutz: your extension IS a, hence when you use the ${EXTEN} variable, you get "a". you need to record the actual extension in a different dialplan variable in order to reference that later in the 'a' extension |
| 20:14.03 | benneton | anyone? Does SIP support groups like ZAP (E.G. SIP/g1) |
| 20:14.07 | InfoNutz | bmoraca: when VoiceMailMain() is called with u${EXTEN} passed it passes the wrong string value, if i hard code the phone number in there pressing * -> voicemailmain(4445556666) it will ask for password and then into the admin menue for that mailbox |
| 20:14.15 | mnt_real | does anyone have digium B410P isdn card ? |
| 20:14.37 | benneton | Does SIP support groups like ZAP (E.G. SIP/g1) |
| 20:15.01 | leifmadsen | no that I'm aware of |
| 20:15.09 | InfoNutz | bmoraca: ah! so $EXTEN is manipulated real time when the user presses * and 'a' is assigned to it... i'll give that a shot |
| 20:15.24 | leifmadsen | ${EXTEN} always is, yes |
| 20:15.39 | [TK]D-Fender | InfoNutz: exten => a,1,VoicemailMain(u${EXTEN}) <--- where the heck does that app say to shove alphabet soup in front of the box NUMBER? |
| 20:15.43 | InfoNutz | thanks i'm new to the whole asterisk platform, very interesting |
| 20:15.44 | leifmadsen | it is always the extension that is left of the => |
| 20:16.13 | [TK]D-Fender | InfoNutz: And the ${EXTEN} is "a" in that case. that is where you "are" |
| 20:16.25 | [TK]D-Fender | InfoNutz: Not what was "dialed" at some point in time. |
| 20:16.32 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 20:16.53 | InfoNutz | i get it now... thanks for the help guys. |
| 20:18.35 | benneton | G u y s . . . Does SIP support groups like ZAP (E.G. SIP/g1) |
| 20:18.41 | [TK]D-Fender | benneton: No |
| 20:19.07 | benneton | I thought so |
| 20:20.13 | flujan | bmoraca: me too but is hard to get help with the hylafax people |
| 20:22.24 | benneton | Thanks [TK]D-Fender ! |
| 20:23.00 | benneton | I'll find something... |
| 20:24.11 | benneton | maybe to use queue? |
| 20:27.10 | Naikrovek | benneton: what are you trying to do |
| 20:27.54 | *** join/#asterisk DrkShadow (n=andrew@host-72-175-240-62.static.bresnan.net) |
| 20:27.55 | *** join/#asterisk servilio (n=servilio@unaffiliated/servilio-ap) |
| 20:27.58 | drclue | All my WiFi phones are actually located on a totally different network than my asterisk. Every step of the way has NAT support and it all supports canreinvite so that media streams can be outsorced to the devices while the control channels remain with asterisk so that server side meadi can be gated in and out of the call |
| 20:29.27 | grandpapadot | drclue: you can only nat once |
| 20:29.55 | rob0 | Opportunity only NATs once. |
| 20:30.15 | DrkShadow | I have a 6757i, I can register fine. I dial, it sends the invite string to the server, and it never gets a response. There is no returned data, not any time I try. Nothing shows up in the error log, and I get nothing from sip debug. |
| 20:30.16 | drclue | Actually I can NAT as many times as I want , but the point is that I can indeed make my NAT work great and retain canreinvite |
| 20:30.36 | grandpapadot | drclue: no, you can only nat once, the second time you nat the source/dest port mapping will be hosed |
| 20:30.41 | [TK]D-Fender | checkout time, BBIAB |
| 20:30.51 | grandpapadot | drclue: you can only nat once from private ip addresses to public |
| 20:31.37 | grandpapadot | drclue: If you're running a subnetted internal network, the only place nat should take place is on your router to the public internet. |
| 20:32.01 | Naikrovek | yes, the router should know of all the networks and route between them |
| 20:32.03 | drclue | Well ,you NAT once (preferred anyway) , but translations can indeed be stacked |
| 20:32.32 | grandpapadot | drclue: Huh? No way man, that's a terrible network design. Why wouldn't you just properly subnet your internal network? |
| 20:32.41 | Naikrovek | stacking has worked for me before but when I realized what I was doing I subnetted correctly |
| 20:33.01 | drclue | I did not say that stacked NAT was a good thing , only that it can be done |
| 20:33.27 | grandpapadot | I can drive down the internstate with my tiny spare tire on too, that would be equally intelligent. |
| 20:33.36 | benneton | Naikrovek - I need to select a non-busy SIP channel from the channel group |
| 20:33.38 | Naikrovek | people do this to segregate their wireless networks from their physical networks for security reasons |
| 20:33.57 | grandpapadot | Naikrovek: It's wrong, no matter what the justification. |
| 20:34.04 | benneton | Naikrovek - just like in Zapata |
| 20:34.07 | *** join/#asterisk ArchGT (n=ArchGT@unaffiliated/archgt) |
| 20:34.11 | Naikrovek | grandpapadot: i'm not saying it's right, just that it happens |
| 20:34.24 | drclue | Look the only point there was it is wrong to say a thing cannot be done when it actually can be done and that the comment should be like some that it is not a good idea |
| 20:34.57 | Naikrovek | benneton: I believe you can set up a ring group within Asterisk, that will ring available phones in the group |
| 20:35.04 | grandpapadot | What I'm saying is that if you're doing it more than once it may be the source of your problem(s) |
| 20:35.13 | Naikrovek | grandpapadot: can't argue there |
| 20:35.18 | drclue | I'm not doing it more than once |
| 20:35.37 | servilio | hi! I am trying to integrate queXS ( quexs.sf.net ) with the asterisk server as configured by eBox ( ebox-platform.com ), but both the Status and the ExtensionState commands behave like there are no channels/extensions; any ideas on why that might be happening? |
| 20:36.37 | benneton | Naikrovek - any link with explaination? |
| 20:36.43 | Naikrovek | benneton: one second |
| 20:36.46 | drclue | If I really needed to do it more than once owing to some silly rules in some system I would , and it can be done , but that is he only point there , it *can* be done. Is it a good thing to do , F**K no , but saying it cannot be done at all is a wrong thing to say |
| 20:37.04 | benneton | Naikrovek - i need to place outgoing call |
| 20:37.23 | benneton | Naikrovek - OK. |
| 20:37.31 | grandpapadot | omg |
| 20:37.44 | Naikrovek | benneton: ah you need to automatically place an outgoing call from an available SIP channel |
| 20:38.23 | benneton | yepp |
| 20:38.31 | Naikrovek | drclue is speaking technically, grandpapadot is speaking practically |
| 20:38.36 | benneton | like with Zap |
| 20:39.14 | drfreeze | Naikrovek: I can't see where it is asking for the file again |
| 20:39.14 | grandpapadot | No, I'm speaking technically, technically and practically it's a wrong implementation of NAT, go read the RFC please. |
| 20:39.14 | Naikrovek | benneton: that's out of my area of expertise, but maybe you can do this with AGI |
| 20:39.14 | drfreeze | Seems like it asked for the wave file only once |
| 20:39.15 | drfreeze | I have turned up logging, but am not seeing the request |
| 20:39.16 | drclue | Naikrovek: you got it |
| 20:39.32 | drclue | FatAGI |
| 20:39.42 | drclue | FastAGI , I should type |
| 20:39.45 | Naikrovek | drfreeze: check the timestamps on the logfiles, usually the phone only uploads its logfile every few days |
| 20:40.01 | benneton | Naikrovek - Will try! Tnx |
| 20:40.11 | Naikrovek | drfreeze: and it's more of a notice than an error; it only plays a welcome tone on the phone when it boots, and only on some phones |
| 20:40.36 | drclue | Benton : FastAGI will allow you to both place a call connected to the current inbound call as well as to originate a call |
| 20:40.52 | Naikrovek | there ya go, benneton, drclue knows |
| 20:42.04 | drclue | I have some nice daemonized bridge code to PHP that will get ya there http://code.google.com/p/fastagi-php-drclue/ |
| 20:42.16 | benneton | drclue - Naikrovek - tnx! You are time savers! |
| 20:42.36 | Naikrovek | its no big deal, someone else would have helped you if i didn't |
| 20:42.38 | Naikrovek | but thanks |
| 20:43.30 | drmessano | Anyone using FAX on an FXS port? |
| 20:43.33 | drclue | Whoever is up to bat can answer a question |
| 20:43.46 | drmessano | On a TDM card |
| 20:44.06 | bmoraca | drmessano: i was for a while, but i'm not anymore |
| 20:44.16 | drmessano | Any quality issues? |
| 20:44.37 | benneton | Naikrovek - show a man how to fish! :) |
| 20:45.00 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 20:45.12 | bmoraca | no, not really. it worked better when it was over a PRI, rather than the PRI-to-SIP gateway I have now...but it still works |
| 20:45.30 | bmoraca | i was able, actually, to dial in to a dialup ISP at about 45kbps |
| 20:47.04 | drclue | I have some PRI's and SIP and all I need to work out is a we bit more on the echo |
| 20:47.39 | drclue | It's almost to the point where nobody cares but me, but I care |
| 20:52.09 | drfreeze | Naikrovek: these are soundpoint 550s |
| 20:52.36 | *** join/#asterisk spck (n=spck@unioncab.com) |
| 20:52.42 | spck | afternoon guys |
| 20:53.08 | spck | anyone have any experience with a redfone fonebridge and connecting it to * via dahdi? |
| 20:56.41 | drclue | So , I'm looking for guinea pigs and general suggestions for my light weight FastAGI php bridge to AGI/AMI . The current offering is hosted as an opensource project at http://code.google.com/p/fastagi-php-drclue/ |
| 20:57.20 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 20:59.04 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:05.47 | drclue | Basically for about 25KB of PHP code you can do almost anything that asterisk has to offer and do so from the common ground of PHP |
| 21:06.47 | drclue | This also affords you the ability to thread out functions that would normally cause wadding issues in asterisk dialplands |
| 21:09.20 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:11.19 | *** join/#asterisk Failrar (n=Failrar@coffee.ipv6.goatse.be) |
| 21:12.26 | *** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel) |
| 21:15.04 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 21:17.16 | *** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net) |
| 21:21.13 | *** part/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 21:23.53 | *** join/#asterisk Skeeter- (n=wil_c_wi@c207.134.244-144.clta.globetrotter.net) |
| 21:26.34 | drfreeze | Naikrovek: Ok, it now downloads the file. I just need to get it to play the ting |
| 21:27.35 | *** join/#asterisk talntid (n=eric@66-208-251-170-Washington.hfc.comcastbusiness.net) |
| 21:35.51 | *** join/#asterisk fun330 (n=manning_@20.190.189.72.cfl.res.rr.com) |
| 21:36.08 | fun330 | where can i find a list of major compaines that use asterisk? |
| 21:36.25 | fun330 | or do any fortune 500 companies use it? |
| 21:41.21 | fun330 | what is bank of america using for their voip system? |
| 21:41.22 | [TK]D-Fender | fun330: Doubt you'll find anything of substance |
| 21:43.41 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 21:44.10 | fun330 | okay, are there any large compaines using asterisk to anyones knowledge? |
| 21:44.27 | KyleK | open source is like being gay right now |
| 21:44.31 | KyleK | don't ask, don't tell |
| 21:44.56 | *** join/#asterisk geneticx (n=geneticx@adsl-149-109-236.mia.bellsouth.net) |
| 21:45.01 | KyleK | actually usually what softare is used is not talked about across the board |
| 21:45.21 | fun330 | haha |
| 21:45.25 | Qwell | software is typically considered a "competitive advantage". |
| 21:45.27 | Qwell | it's odd. |
| 21:45.39 | Qwell | and no, I'm not kidding. |
| 21:45.43 | fun330 | okay so i am not going to find anything then huh? |
| 21:45.53 | geneticx | any recommendations for wireless DECT phones that have 'on hold' ? |
| 21:46.03 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:46.24 | *** join/#asterisk LiNeTuX (n=LiNeTuX@119.153.205.68.cfl.res.rr.com) |
| 21:46.30 | drclue | fu330:Asterisk is so transparent that most times it is had to tell. The act is that Asterisk is the de facto standard , much as Apache is the De facto standard for web serving |
| 21:47.37 | Qwell | fun330: start here. http://www.digium.com/en/company/casestudies/ |
| 21:47.38 | fun330 | yeah i am trying to get a big client but they are nervious about using asterisk and they just want to see that other companies use it |
| 21:48.56 | drclue | Well , fun330 , not sure what to tell ya except that you would be hard pressed to find similar software beyond Asterisk that has any market share worth mentioning |
| 21:50.08 | *** join/#asterisk errotan (n=errotan@62.201.123.198) |
| 21:50.30 | fun330 | well they are looking at going with shoretel |
| 21:50.38 | fun330 | but the quote was crazy |
| 21:50.48 | fun330 | i don't know why they would even consiter it |
| 21:51.22 | drclue | Fine , have shoretel cite their user base and then compare that to asterisk. I don't even need to look it up |
| 21:52.01 | drclue | Often name brand PBX systems are (while perhaps not in this case) Asterisk under the hood |
| 21:53.48 | drclue | Asterisk is like Apache. Supported and maintained be the masses. Most web servers are Apache and not some slick named private offering and the world runs on it. Asterisk is in it's realm the same thing as Apache |
| 21:54.37 | drclue | If your client is stuck on stupid , so be it. Rake em for some good cash |
| 21:58.22 | bmoraca | an asterisk system is only as strong as its integrator. i can understand them being wary. |
| 21:59.06 | bmoraca | and asterisk is not comparable to apache. for one, apache has been around a LOT longer, and for two, Asterisk is not the de facto standard in IP telephony. big difference. and asterisk has a long way to go before it reaches that status. |
| 21:59.42 | drclue | bmoraca: Most punters would not know asterisk from timex. It is up to the consultant to make the case |
| 21:59.44 | [TK]D-Fender | And hopefully telephony will be deprecated by telepathy :p |
| 22:00.17 | drclue | Thankyou SPMender |
| 22:02.45 | drclue | bmoraca: who would you say *is* the defacto standard? |
| 22:04.10 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-169-241.dsl.stlsmo.sbcglobal.net) |
| 22:06.20 | drclue | I'm a 30+ year veteran programmer transitioning into VOIP and there seems to be ZERO other viable contenders in this space, but hey if you know something I missed , I want to here about it |
| 22:07.52 | drclue | bmoraca : not trying to dis ya . but facts is facts. Either your wrong or I am. Plese set me straight |
| 22:07.59 | bmoraca | drclue: it's all about the names that people recognize: Shoretel, Nortel, Cisco, Avaya, etc. The people that sign the checks and ultimately make the decisions have probably never heard of Asterisk. That's the difference. |
| 22:08.33 | fun330 | yeah i agree with that, i am having that problem right now |
| 22:08.47 | LemensTS | so asterisk uses slin by default, so if i record a message in g729 and play it in a g729 channel using Background it shouldnt decode it but it does. here is a cli output of what its doing http://pastebin.com/m2c091aae |
| 22:08.50 | bmoraca | drclue: i've run into this many times. people are reticent about a product they've never heard of, and it's not always possible to sway them, despite the cost savings. |
| 22:08.59 | KyleK | huh |
| 22:09.05 | drclue | bmoraca: piss on that. Of course if the punter wants to pay ya , go for it , but I sorta thought we were talking about what is mainstream |
| 22:09.19 | LemensTS | u can see there is 0 license in use, then it does the background command, and there is 1 license in use |
| 22:09.36 | *** join/#asterisk errotan (n=errotan@62.201.123.198) |
| 22:10.01 | LemensTS | I do see the extensions is hung up (its a originate cmd) |
| 22:10.02 | KyleK | i saw a mention of wideband codecs on the mailing list, how common is the support of wide band? |
| 22:10.06 | bmoraca | drclue: and how do you define "mainstream"? most people would define it as "who has the largest install base", and I can GUARANTEE you that asterisk is very far down on that list (although they are climbing) |
| 22:10.11 | drclue | Sometimes it is a test. Stand your ground and get paid |
| 22:10.45 | drclue | bmoraca: with all due respect , ya full of shit. |
| 22:11.08 | KyleK | LemensTS: hrm that log proves its decoding g729 but can you get information on what codec is in use? |
| 22:11.36 | bmoraca | whatever. you've been spouting shit all day in here. i wouldn't be surprised if you don't have a single commercial installation under your belt. until you've been out there trying to sell it for a while, you've got no clue. |
| 22:11.54 | drclue | whoes shit I don't know , but bettr than 90% of VOIP is via Aserisk , unless of course ya want to *PROVE* me wrong |
| 22:12.07 | *** join/#asterisk PY8AZT (n=PY8AZT@201009185039.user.veloxzone.com.br) |
| 22:12.59 | bmoraca | 90% of voip is asterisk? what the hell are you smoking? troll. |
| 22:13.01 | KyleK | LemensTS: can you "sip show channels" during that Background() as well? |
| 22:13.19 | KyleK | heehee reverse trolling? |
| 22:13.34 | drclue | bmoraca: I'm always open to new knowledge , so state your facts |
| 22:13.58 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 22:14.27 | KyleK | ~wb |
| 22:14.28 | infobot | It's great to be back! |
| 22:14.32 | KyleK | ~wideband |
| 22:14.41 | KyleK | aww cmon throw me a bone |
| 22:15.22 | outtolunc | ~g722 |
| 22:15.23 | infobot | somebody said g722 was a high bit rate (48/56/64Kbps) ITU standard codec. |
| 22:15.34 | *** join/#asterisk phuff (n=user@sourceforge/staff/luapffuh) |
| 22:15.54 | drclue | bmoraca: I'm listening for my education? |
| 22:16.10 | phuff | If I wanna setup a command like *9 so that it sends things directly to voicemail until I dial *9 again... |
| 22:16.20 | phuff | Does that mean I have to reload the dialplan somehow? |
| 22:16.27 | fun330 | here is a good article about open source pbx http://www.nojitter.com/showArticle.jhtml?articleID=212903167 |
| 22:16.31 | KyleK | saying 90% of voip is asterisk sounds very factless |
| 22:16.42 | KyleK | so proving it right or wrong is probably non-trivial |
| 22:17.40 | grandpapadot | phuff: *9,1,Set(DB(family/key)=${foo}) |
| 22:17.47 | grandpapadot | phuff: then use ExecIf() and evaluate $foo to see what needs to happen |
| 22:17.52 | phuff | Ah |
| 22:17.58 | phuff | So _that's_ what you use the DB for :) |
| 22:18.06 | LemensTS | KyleK: 0x100 (g729) says that the whole time |
| 22:18.06 | drclue | bmoraca: please ,educate me about this more common than asterisk thing I need to support! tick tick ick tick...bullshit |
| 22:18.13 | phuff | grandpapadot: Thanks. |
| 22:19.08 | KyleK | huh sounds like a bug or somethings wrong with your file |
| 22:19.17 | grandpapadot | phuff: no problem, md |
| 22:19.38 | drclue | bmoraca: come on girlfriend type faster, type something , tick , tick tick ,tick ...silence = bullshit |
| 22:20.18 | KyleK | lol now you're trolling :) |
| 22:21.01 | Naikrovek | drfreeze: let me know if you get it to play, my phones wont play it. something in sip.cfg i'm sure. |
| 22:21.22 | KyleK | damn spa3102 doesn't do g722 |
| 22:21.29 | drclue | Asterisk *IS* the number one telephony system , no holds barred and without proof that it aint the Apache of telephone , your just talking shit |
| 22:22.06 | grandpapadot | Well guys, I've had enough romper room, later all. |
| 22:22.11 | KyleK | why are you trying to convince us? |
| 22:22.24 | geneticx | later grandpapadot. |
| 22:22.32 | KyleK | people to tell about asterisk: people that are NOT in #asterisk ;) |
| 22:22.57 | drclue | I'm just responding to bmoraca , as I know most san folks here know at least that much if *nothing* else |
| 22:22.57 | *** join/#asterisk galeras (n=galeras@186.80.181.115) |
| 22:24.22 | *** join/#asterisk Defraz (n=T0tal@corp.fuzecore.com) |
| 22:26.16 | cusco_ | can't figure what is wrong with pri: http://paste.debian.net/44519/ |
| 22:26.21 | cusco_ | that happens when I try to outbound |
| 22:26.34 | cusco_ | it goes unreacheble, calls comming in cannot reach asterisk neither |
| 22:26.38 | cusco_ | any hints |
| 22:26.39 | cusco_ | ' |
| 22:29.13 | cusco_ | Channel 0/1, span 1 got hangup request, cause 47 |
| 22:29.19 | cusco_ | Auto fallthrough, channel 'SIP/311-54001440' status is 'CHANUNAVAIL' |
| 22:32.11 | galeras | Please: Which softphones supports URL dial parameter? |
| 22:33.08 | bmoraca | x-lite and eyebeam do, i believe |
| 22:35.27 | LemensTS | Which softphones support g729 free? |
| 22:35.44 | galeras | bmoraca: no mf, i'm trying with eyebeam wihout success! |
| 22:35.46 | LemensTS | Sippax says it does but it diddnt |
| 22:36.03 | LemensTS | xlit dont |
| 22:36.14 | KyleK | I don't think any of the free ones do |
| 22:36.33 | galeras | LemensTS: So far , only payed softphones do |
| 22:37.00 | *** join/#asterisk M_Red (n=ty@97-119-253-80.spkn.qwest.net) |
| 22:37.01 | Qwell | there is no so far.. |
| 22:37.13 | Qwell | You have to pay patent royalties. |
| 22:37.31 | KyleK | anyone know when the patents expire? :) |
| 22:39.02 | Qwell | 2014? |
| 22:39.23 | Qwell | wait, no, that's G.723.1 |
| 22:39.34 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 22:40.48 | Qwell | coppice or Corydon76-dig would likely know |
| 22:41.54 | *** part/#asterisk M_Red (n=ty@97-119-253-80.spkn.qwest.net) |
| 22:43.38 | *** join/#asterisk rossand (n=rossand@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 22:44.02 | phuff | is NoOp cheaper than Verbose? |
| 22:44.11 | Qwell | phuff: They do different things. |
| 22:44.16 | Qwell | NoOp does nothing. At all. |
| 22:44.21 | phuff | But it prints, right? |
| 22:44.39 | Qwell | NoOp? No. It does nothing. |
| 22:44.39 | russellb | printing is a side effect, done by the PBX core, not within the app itself |
| 22:44.56 | Qwell | You'll see an output line if you use Verbose. |
| 22:45.10 | phuff | interesting |
| 22:45.12 | Qwell | (in addition to the execution of the priority in the dialplan by the PBX) |
| 22:45.45 | phuff | So, if I just wanna print something to the console in a dialplan, best to use NoOp or Verbose ? |
| 22:46.03 | Qwell | Verbose, since it actually prints something. |
| 22:46.12 | phuff | Ok |
| 22:46.22 | LemensTS | KyleK: just bought eyebeam: called from eyebeam g729 out itsp via g729 with no decoding..., called from eyebeam g729 into asterisk and played a g729 recorded file with no decoding....But when asterisk originates a call out that itsp, and connects it with a phpagi script it transcodes when it hits BACKGROUND cmd. |
| 22:46.29 | phuff | Weird, I'm seeing NoOp in a lot of example dialplans to try and put things in console |
| 22:46.35 | phuff | At least, that's what it looks like |
| 22:46.38 | Qwell | phuff: They are wrong. :) |
| 22:46.39 | phuff | NoOp(${variable}) |
| 22:46.42 | phuff | and such |
| 22:46.44 | Qwell | That's simply a side-effect. |
| 22:46.46 | *** part/#asterisk Skeeter- (n=wil_c_wi@c207.134.244-144.clta.globetrotter.net) |
| 22:47.35 | Qwell | NoOp() doesn't even technically take any arguments. |
| 22:48.13 | phuff | So did it used to be the preferred method of printing things to console or something? |
| 22:48.38 | Qwell | well, it only shows up if you have dialplan output displayed. most people do |
| 22:48.51 | phuff | Ah |
| 22:48.52 | phuff | I see |
| 22:48.58 | phuff | so Verbose is technically correct because it'll always display |
| 22:49.03 | phuff | But NoOp there's no guarantee? |
| 22:49.28 | Qwell | well, it won't always display either. depends on verbosity level |
| 22:49.47 | Qwell | </pedantic> |
| 22:49.48 | Qwell | use either. |
| 22:52.49 | phuff | Hah |
| 22:52.58 | phuff | Yeah, I see what you're saying about the verbosity level |
| 22:53.05 | phuff | But the idea is, Verbose() is designed for output |
| 22:53.07 | phuff | NoOp isn't |
| 22:53.15 | phuff | Verbose is more semantically correct if I want to put something to the console |
| 22:54.07 | phuff | Is there some way of seeing the output of an AGI for debugging? |
| 22:54.15 | phuff | Like if it fails somewhere? |
| 22:55.19 | galeras | Please, suggestme a softphone with support for URL dial parameter. Eyebeam and Zoiper (free) seems do not. |
| 22:57.53 | *** join/#asterisk raden_work (n=tanning@69-179-99-17.stat.centurytel.net) |
| 22:58.13 | raden_work | is there anything i need to reload when i edit voicemail.conf ? |
| 23:00.16 | raden_work | [Aug 18 17:56:03] WARNING[14436]: app_voicemail.c:7605 vm_authenticate: Couldn't read username |
| 23:01.07 | *** join/#asterisk lmsteffan (n=laurent@reef.ac-noumea.nc) |
| 23:01.08 | raden_work | <PROTECTED> |
| 23:01.37 | raden_work | 101 => 4444,Jon,**email** |
| 23:03.10 | Qwell | Did you reload? |
| 23:04.48 | raden_work | sip and dial plan |
| 23:04.54 | Qwell | and voicemail? |
| 23:05.06 | raden_work | how do i reload voicemail ? |
| 23:05.17 | raden_work | duh nevermind |
| 23:05.25 | raden_work | type help and i shall see |
| 23:07.01 | *** part/#asterisk drclue (n=drclue@ip-65-49-160-6.wireless.dyn.beamspeed.net) |
| 23:13.14 | *** join/#asterisk davidandgoliath (n=David@bowness-out.eng.telusmobility.com) |
| 23:13.50 | raden_work | how would i make a extension to dial a outside number if it wasnt picked up at my desk phone ? |
| 23:13.55 | raden_work | exten => 101,n,Dial(SIP/101,20) |
| 23:14.12 | raden_work | after that i would like it to try a cell phone for 15 seconds |
| 23:18.34 | *** join/#asterisk voxter (n=voxter@74.173.119.66.host.metrobridge.net) |
| 23:20.37 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 23:22.37 | LemensTS | what does g729:60 mean? |
| 23:26.59 | raden_work | 60:1 dilution |
| 23:27.35 | raden_work | 60 Bytes |
| 23:29.50 | *** join/#asterisk ming_zym (n=ming_zym@210.192.100.184) |
| 23:31.58 | *** join/#asterisk LiNeTuX (n=LiNeTuX@119.153.205.68.cfl.res.rr.com) |
| 23:32.45 | raden_work | how can i dial out via the dial plan like a cell phone 8885554444 ? |
| 23:34.40 | LemensTS | exten => _X.,1,Dial(SIP/${EXTEN}) |
| 23:35.03 | LemensTS | exten => _X.,1,Dial(SIP/context/${EXTEN}) |
| 23:38.26 | raden_work | got that working |
| 23:38.56 | raden_work | is there a way to turn parts of the dial plan on and off ? |
| 23:39.14 | raden_work | like pickup my phone and make it so it dont forward to my cell and just go direct to voicemail |
| 23:39.57 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 23:48.17 | phuff | raden_work: grandpapadot said to use something like this: *9,1,Set(DB(family/key)=${foo}) |
| 23:48.17 | phuff | <PROTECTED> |
| 23:48.30 | phuff | and then use ExecIf |
| 23:48.52 | raden_work | i better keep reading |
| 23:51.46 | *** join/#asterisk PuroOsso (n=PuroOsso@187.21.17.186) |
| 23:52.21 | Katty | :> |
| 23:52.34 | Katty | HEWWOES |
| 23:59.17 | *** join/#asterisk davidandgoliath (n=David@bowness-out.eng.telusmobility.com) |