| 00:00.48 | *** join/#asterisk [Jasper] (n=jverberk@195-240-174-59.ip.telfort.nl) |
| 00:00.55 | [Jasper] | hello people, I have a question about asterisk |
| 00:01.01 | [Jasper] | and a problem I'm running into |
| 00:01.34 | beek | ~ask |
| 00:01.35 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 00:01.53 | [Jasper] | I made a standard sip trunk and I'm trying to dial-in to that number,...I also defined context and stuff....weird thing is....when I receive a call, it doesn't go to the correct context...all it does is go to default |
| 00:02.18 | [Jasper] | when i include the correct context in the default context...my phone rings |
| 00:02.34 | [Jasper] | but I want it to go to the correct one immediately |
| 00:02.39 | manxpower | [Jasper]: then the incoming call is not authenticating as who you think it is |
| 00:02.39 | [Jasper] | instead of through some dirty trick |
| 00:02.50 | manxpower | if it authenticated that way then it would work |
| 00:03.04 | [Jasper] | ok, how can I see how it is authenticated? |
| 00:03.11 | manxpower | That is the CLASSIC auth issue |
| 00:03.41 | manxpower | if it goes to the wrong context |
| 00:03.41 | [Jasper] | what is the classic solution? |
| 00:03.42 | [Jasper] | :P |
| 00:04.01 | manxpower | pastebin the output of a failed call |
| 00:04.48 | [Jasper] | k |
| 00:06.54 | [Jasper] | manxpower http://pastebin.com/m142ce826 |
| 00:07.40 | manxpower | SIP/80.252.84.190-0854ffc0 <-- it didn't auth as anyone |
| 00:08.05 | manxpower | [thestuffinhere] would be listed instead of the IP if it had authed at that user |
| 00:08.49 | manxpower | are you using 1.4 or 1.6? |
| 00:10.05 | [Jasper] | 1.4 |
| 00:10.27 | *** part/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 00:10.34 | manxpower | 1.6 has an allowguest=no IIRC |
| 00:10.46 | [Jasper] | so I should upgrade? |
| 00:11.07 | manxpower | Here's a good description of the Polycom WiFi phone. "In an office of 16 VoIP geeks, the polycom WiFi phone is sitting in a box of junk in a closet." |
| 00:11.29 | manxpower | [Jasper]: all that will do is make your unauthed calls fail |
| 00:11.46 | manxpower | rather than the calls being accepted and matching the settings in [general] |
| 00:12.21 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 00:12.49 | beek | evening [TK]D-Fender |
| 00:13.03 | [TK]D-Fender | beek: evening.. |
| 00:13.20 | [Jasper] | hmm |
| 00:13.25 | [Jasper] | so what should I do manxpower ? |
| 00:14.00 | *** join/#asterisk Dovid (n=annon@tony09-121-90.inter.net.il) |
| 00:14.18 | manxpower | [Jasper]: figure out why it's not authing or authing as the wrong user |
| 00:15.08 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 00:15.58 | bmoraca | manxpower: wifi phones = lol |
| 00:15.59 | [Jasper] | hmm |
| 00:16.40 | [Jasper] | manxpower could it be in my signing in |
| 00:16.41 | [Jasper] | http://www.budgetphone.nl/forum/viewtopic.php?t=46 |
| 00:16.45 | [Jasper] | the register being different? |
| 00:23.45 | [TK]D-Fender | [Jasper]: register => <number>@budgetphone.nl:<password>@sip.budgetphone.nl /<number> |
| 00:23.45 | citywok | according to the wiki you need to compile asterisk to get CDR to be able to log the uniqueid, is this true, or is there some other reason it wont log it (that note is for asterisk 1.2, using 1.4.22 now) |
| 00:24.40 | [TK]D-Fender | [Jasper]: without the spaces at the end |
| 00:24.43 | *** join/#asterisk johnakabean (n=none@pool-72-82-113-23.nrflva.east.verizon.net) |
| 00:24.45 | icel | is there a way to figure out how many concurrent calls are happening in * ? |
| 00:25.01 | bmoraca | icel: core show channels |
| 00:25.06 | [TK]D-Fender | icel: "core show channels concise" |
| 00:25.17 | icel | thx |
| 00:25.42 | codefreeze-lap | citywok: uniqueid isn't always unique. Just remember that. Some drivers allow you to configure some fields |
| 00:26.25 | johnakabean | hey fender, thanks for help yesterday; i went back to 1.4 though as mpg 123 wouldn't work; but, i started having a major problem with 1.6. Remember when I asked if yours beeped when you shut it down? For some reason asterisk 1.6 was a run away! It wouldn |
| 00:26.32 | johnakabean | t stop uhmm starting up |
| 00:26.34 | johnakabean | lol |
| 00:26.46 | [Jasper] | [TK]D-Fender where did you get that budgetphone intel? |
| 00:28.36 | [TK]D-Fender | johnakabean: Meh... can't account for the rest, but it was a far better try this time |
| 00:30.08 | *** join/#asterisk km2 (n=x@cpe-74-64-12-212.nyc.res.rr.com) |
| 00:36.58 | *** join/#asterisk dr0ck (n=dr0ck@nat/digium/x-0e0a53dab05844e7) |
| 00:51.10 | DarkRift | [TK]D-Fender, you know any good voip supplier that ships to the province of Quebec with good prices for both phones and asterisk hardware ? |
| 00:52.38 | [TK]D-Fender | DarkRift: Before the exchange rate got smacked, Telephonydepot was great, even on import |
| 00:53.04 | DarkRift | Still a good choice ? |
| 00:53.44 | [TK]D-Fender | DarkRift: try : http://www.canadianvoipstore.com/home.php |
| 00:54.02 | DarkRift | Alright I'll check those 2, thanks |
| 00:54.03 | [TK]D-Fender | DarkRift: decent price and IIRC ships in reasonably without duty |
| 00:56.35 | *** join/#asterisk MaliutaLap (n=biteme@203.171.192.132) |
| 00:57.16 | *** join/#asterisk IvanG (n=IvanG@78.52.238.127) |
| 01:01.36 | *** join/#asterisk masteryoda (n=matthew@adsl-163-40-99.hsv.bellsouth.net) |
| 01:01.37 | [TK]D-Fender | DarkRift: Clearly the exch rate has indeed hit them too |
| 01:05.33 | *** join/#asterisk riddlebox (n=victoria@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:09.24 | johnakabean | I just don't understand why asterisk 1.6 would be picky about mpg123 while 1.4 wasnt; they didn't change the sound codecs so it should recognize the stream. |
| 01:11.03 | johnakabean | As far as asterisk being runaway, I'm sure its an error on my part |
| 01:11.12 | beek | GN all |
| 01:11.22 | [TK]D-Fender | beek: Nite |
| 01:14.40 | *** join/#asterisk obnauticus (n=lol@about/windows/regular/obnauticus) |
| 01:15.07 | *** join/#asterisk sosoriri (n=sosoriri@218.207.141.90) |
| 01:15.17 | johnakabean | -- Remote UNIX connection it would be nice if it would provide the ip address |
| 01:15.29 | johnakabean | in case it was a security issue |
| 01:16.58 | [TK]D-Fender | johnakabean: Your GUI monitoring scripts are h4x0ring you! |
| 01:18.26 | *** join/#asterisk keebler (n=keebler@h199.233.20.98.dynamic.ip.windstream.net) |
| 01:18.48 | johnakabean | no i allow 3 other people that use a2billing to login to console |
| 01:19.05 | johnakabean | but I would like to be able to see if they give out their password |
| 01:19.24 | johnakabean | based on ip address and their current location |
| 01:19.48 | *** join/#asterisk fun330 (n=manning_@169.165.8.67.cfl.res.rr.com) |
| 01:19.56 | johnakabean | the gui scripts say manager logged on and off from 127.0.0.1 |
| 01:19.57 | *** part/#asterisk fun330 (n=manning_@169.165.8.67.cfl.res.rr.com) |
| 01:20.05 | johnakabean | but when they connect it just says remote unix connection |
| 01:20.15 | johnakabean | they onlly installed the asterisk console, not asterisk itself |
| 01:21.08 | johnakabean | they only use the asterisk console |
| 01:21.19 | johnakabean | i have on their shell login a script to open the asterisk console |
| 01:21.30 | johnakabean | and deny everything else |
| 01:21.33 | citats | johnakabean: if you want to see the ip they connect with check your ssh/telnetd logs |
| 01:21.33 | Nugget | telnet is eeeeeeevil! |
| 01:21.36 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:21.47 | johnakabean | yeah, citats, but you have to take the tiime to cross reference |
| 01:22.03 | johnakabean | if it was emergency i would be fuxed |
| 01:22.48 | citats | johnakabean: asterisk doesn't know anything about IP address they are connecting from. its just from a socket. if you want to grab that info at the same time make their shell a script that also records that somewhere |
| 01:22.58 | johnakabean | i HAVE had my asterisk box hacked before and they made 1000 calls a minute to some home shopping network, created a conference, and put all the operators answering in it. |
| 01:23.33 | johnakabean | wasn't fun to have 2023243000 call me |
| 01:23.40 | johnakabean | fbi in washington dc |
| 01:24.09 | johnakabean | that's why my firewall is STRICT |
| 01:24.23 | johnakabean | only allows sip, rtp, and iax2 packets from my trunks |
| 01:24.35 | johnakabean | and 5038 is DENIED |
| 01:25.23 | johnakabean | i can hear you guys laughing now |
| 01:25.31 | johnakabean | i thought it was funny until i got the call |
| 01:26.02 | *** join/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis) |
| 01:26.04 | *** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 01:26.04 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:26.34 | nix8n82 | it still is |
| 01:26.57 | johnakabean | yeah well it was some guy in austrailia (based on ip) |
| 01:27.03 | johnakabean | i had to surrender my logs |
| 01:27.20 | *** join/#asterisk riddlebox (n=victoria@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:27.25 | nix8n82 | did they find him? |
| 01:27.32 | johnakabean | i don't know but they left me alone |
| 01:27.58 | johnakabean | they are watching my ip though |
| 01:28.08 | nix8n82 | cool..so you only had to buy one new pair of underware |
| 01:28.10 | nix8n82 | ? |
| 01:28.14 | johnakabean | yeah |
| 01:28.26 | johnakabean | if i didn't surrender the logs i was going to get the charges |
| 01:28.47 | nix8n82 | yeah no shit..I would of done the same |
| 01:29.11 | johnakabean | they were charging 5 cents a minute for the 800 number's charges and 40 counts of 'annying ringing" class 3 felony |
| 01:29.43 | johnakabean | i was looking at 5 years max and 30000 dollars |
| 01:30.05 | nix8n82 | that would suck |
| 01:30.14 | citats | if they just called you on the phone you werent looking at anything aside from intimidation |
| 01:30.41 | johnakabean | well i think they actually did try to find me at first |
| 01:30.55 | johnakabean | but my ip shows up in arpa 500 miles from where i'm actually at |
| 01:31.17 | johnakabean | bad networking on verizons part |
| 01:31.31 | johnakabean | and i don't use the modem verizon gave me so they couldn't link the mac to my account |
| 01:32.21 | nix8n82 | broadband? |
| 01:32.35 | citats | thats not true at all. otherwise all anybody needs to do to get free service is just grab a random modem |
| 01:32.36 | johnakabean | i use a Dslplus bridge i bought online and put it in ppoe.....verizon does NOT require you to put in your right username and password to login to the ppoe server.. you just have to pu 3 characters on each |
| 01:33.00 | johnakabean | no, citats, they have to put the dsl signal on your line from the ATM at the central office |
| 01:33.20 | johnakabean | I have dry loop |
| 01:33.28 | johnakabean | no phone just the dsl signal |
| 01:33.35 | citats | oh thats right, i know nothing about this. forget i brought it up |
| 01:33.49 | johnakabean | ??? |
| 01:34.16 | johnakabean | verizon didn't own this network to start with, it was a small phone company at first |
| 01:34.30 | johnakabean | so when they switched over, they were lazy at their logging tactics |
| 01:35.01 | johnakabean | I'm 1000 feet of cable from the central office; I could throw a rock at it |
| 01:35.24 | johnakabean | i get max speed and 98% quality of service |
| 01:35.27 | johnakabean | 1 ms jitter |
| 01:35.54 | johnakabean | they're running fiber lines now for fios :) |
| 01:36.20 | johnakabean | they're done in my part of town but they havent' finished elsewhere so they make us wait |
| 01:36.21 | nix8n82 | anyone know of obvious dis advantages to using phpagi for agi scripts? especially if the same agi is going to be called 500 time or so in any given second. for 500 different calls? |
| 01:36.45 | johnakabean | agi of asterisk? |
| 01:37.17 | johnakabean | i'm sure they will make the fios network traceable |
| 01:37.33 | nix8n82 | yeah that would be the one..I have a couple perl scripts but I want to convert them to php |
| 01:37.34 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 01:38.32 | citats | nix8n82: if your using normal agi then you have the startup cost of your agi. which can be quite high with an interpreted language like php or perl. a compiled program will be quicker, or you could use fastagi |
| 01:39.10 | [TK]D-Fender | 500 calls per second alone is psycho. AGI on top? |
| 01:39.11 | johnakabean | i only deal with php; i do know running php independtly of the service using it causes more resource usage versus running it as a module. this is experience from apache. |
| 01:39.12 | [TK]D-Fender | ~wglwat |
| 01:39.12 | jbot | i guess wglwat is well, good luck with all that |
| 01:39.42 | Qwell | well, using fastagi there shouldn't be a hell of a lot of overhead |
| 01:40.11 | johnakabean | i have used two php scripts with asterisk that's it |
| 01:40.26 | Qwell | ahh, citats said normal agi though. nm |
| 01:41.00 | Qwell | I doubt phpagi works with fastagi, so...there's one major disadvantage |
| 01:41.12 | Qwell | if you're going to be doing anywhere near that volume, you definitely want fastagi |
| 01:41.26 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 01:41.36 | johnakabean | on that note qwell, he could use fastagi to execute php as part of the OS but that possible would be a security issue |
| 01:41.48 | Qwell | how? |
| 01:42.25 | johnakabean | agi is basically commandline execution |
| 01:42.41 | Qwell | okay? |
| 01:42.50 | nix8n82 | well about 200 to 250 calls open at anyone time |
| 01:42.59 | nix8n82 | not really per second |
| 01:43.02 | johnakabean | if he has asterisk running as root |
| 01:43.06 | Qwell | nix8n82: either way |
| 01:43.25 | Qwell | johnakabean: don't do that then? |
| 01:43.28 | johnakabean | well, nix, the calls wouldn't use the agi the whole time, just when they got to that part of the dialplan |
| 01:43.43 | johnakabean | no qwell, otherwise your agi scripts will run as root |
| 01:44.03 | Qwell | so whats the problem? |
| 01:44.13 | Qwell | Don't run Asterisk as root |
| 01:44.15 | johnakabean | never heard of php injection |
| 01:44.26 | nix8n82 | Why fastagi if it's on one server? |
| 01:44.27 | Qwell | don't have poor coding skills either |
| 01:44.40 | Qwell | nix8n82: fastagi doesn't have to spawn php every time |
| 01:44.50 | Qwell | fastagi just stays open and handles socketed requests |
| 01:45.13 | nix8n82 | so would it create an instance for each channel? |
| 01:45.25 | nix8n82 | because I'm having the script collect dtmf data |
| 01:45.40 | johnakabean | storing in mysql, nix? |
| 01:45.52 | nix8n82 | yeah |
| 01:47.12 | johnakabean | anyway to have asterisk use its connection to store outside of asterisk's databases? |
| 01:47.32 | johnakabean | in mysql? If so i would use that otherwise you would have mysql overhead. |
| 01:47.45 | nix8n82 | so having an agi being called lets say 15 times for one call over an 10 min period would be a huge burden to the machine? |
| 01:48.05 | johnakabean | hell yeah |
| 01:48.17 | johnakabean | lol 15 times? why not collect all you need first then initiate the storage |
| 01:48.35 | Qwell | nix8n82: depends on the frequency. it can, yes |
| 01:48.51 | johnakabean | he just said 15 times PER CALL qwell |
| 01:48.57 | nix8n82 | I wouldn't write to mysql 15 times. |
| 01:49.06 | johnakabean | oh possibly no then |
| 01:49.23 | johnakabean | you use php to validate it those other 14 right? |
| 01:49.57 | nix8n82 | or store my results in a file then have another gather the data |
| 01:51.20 | *** join/#asterisk cp5 (n=samy@cpe-76-171-169-53.socal.res.rr.com) |
| 01:51.21 | johnakabean | clear the file between calls |
| 01:51.23 | nix8n82 | so I can keep sql traffic to a min |
| 01:51.25 | johnakabean | just reminding |
| 01:51.41 | nix8n82 | yeah and each file would be unique |
| 01:51.52 | cp5 | hi guys. what's the reason for the "jbforce" flag with the jitterbuffer? for example, in chan_sip, just enabling the jitter buffer isn't enough, you must force it too. why are there two flags? |
| 01:52.32 | cp5 | fyi, the changelog for 1.6.0.4-rc1 in http://www.asterisk.org/node/48561 points to a 404 |
| 01:53.11 | Qwell | samy is my hero |
| 01:53.22 | *** join/#asterisk fun330 (n=manning_@169.165.8.67.cfl.res.rr.com) |
| 01:53.42 | Qwell | cp5: They've got you working on Asterisk stuff now? :p |
| 01:53.53 | cp5 | what's up qwell! how are you? |
| 01:53.57 | fun330 | what is the best way to secure an asterisk server plugged right into the public interent |
| 01:54.05 | cp5 | i've worked on asterisk stuff for a long time man! |
| 01:54.18 | Qwell | cp5: just trolling, heh |
| 01:54.29 | cp5 | nice nice |
| 01:54.34 | Qwell | I was leading up to a "man, you need more engineers..." joke. :P |
| 01:54.38 | cp5 | hah |
| 01:54.56 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:55.07 | johnakabean | ok, the variables for asterisk are arg1, arg2, arg3 right? |
| 01:55.18 | johnakabean | do you set those in php or ? |
| 01:55.25 | Qwell | cp5: trying to remember the reason for that though |
| 01:56.02 | johnakabean | or can you make custom variables already in your php scripts |
| 01:56.05 | johnakabean | for asterisk |
| 01:56.21 | johnakabean | or is it just limited to arg1, arg2, arg3, and so on |
| 01:56.29 | cp5 | Qwell, yeah, i assume some other channel may behave differently with and without force, but i wouldn't know why. on SIP there just seems little point for the force flag, and the enable flag is just teasing me |
| 01:56.36 | Qwell | cp5: is this 1.2 or something? |
| 01:56.46 | cp5 | qwell, 1.6.0 |
| 01:56.47 | Qwell | no, nm |
| 01:57.29 | Qwell | I think it had something to do with which side the jitter was on...or something |
| 01:58.01 | Qwell | yeah, the comments in the sample config explain it |
| 01:59.37 | cp5 | i'm looking at the sip.conf example comments and am confused by what it's saying |
| 02:01.23 | cp5 | to me it seems to be saying "you must have jbenable and jbforce in order for the jitter buffer to work". also the line "An enabled jitterbuffer will be used only if the sending side can create and the receiving side can not accept jitter" is a bit confusing to me -- i assume it means "an enabled jitterbuffer will be used if the sending side can create [jitter] and the receiving side [asterisk] can not accept jitter [what does it technically |
| 02:01.43 | cp5 | what i don't understand is why you would ever have jbenable=yes but jbforce=no |
| 02:02.32 | Qwell | well, think of different channel types |
| 02:02.46 | Qwell | if you've got a channel in and a channel out, both sip, if there's jitter, it's fine |
| 02:03.26 | Qwell | so, if you just enable it, nothing is going to happen unless you force it |
| 02:03.36 | Qwell | now consider a sip channel and a zap channel |
| 02:03.46 | Qwell | something like that |
| 02:03.49 | cp5 | why would that be ok though? what is the reason for enabling it in the first place if it will never kick in |
| 02:04.07 | Qwell | chan_sip doesn't know or care where the destination is |
| 02:04.58 | cp5 | so you're saying it shouldn't account for jitter if it's just passing traffic? |
| 02:05.08 | Qwell | something like that |
| 02:05.25 | Qwell | this is all from memory, and probably not entirely accurate |
| 02:05.36 | file | it's... the way the person who wrote it implemented the jitterbuffer... it's... yeah |
| 02:08.31 | cp5 | it just seems there's no point to the force option then, you either want it on or not |
| 02:09.34 | file | my mental capacity is not at the level required to comment on such a thing |
| 02:10.32 | johnakabean | cp, what distro of linux? |
| 02:10.56 | *** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk) |
| 02:17.58 | *** join/#asterisk coppice (n=chatzill@201.193.17.210.dyn.pacific.net.hk) |
| 02:19.45 | cp5 | johnakabean, me? |
| 02:20.00 | Qwell | cp5: he's dumb, ignore him |
| 02:20.08 | cp5 | :\ |
| 02:20.28 | cp5 | qwell you ever come out to LA? |
| 02:20.40 | Qwell | yeah.. didn't jjshoe tell you about it? |
| 02:21.07 | Qwell | y'all didn't communicate, and the hotel screwed me over pretty nicely :p |
| 02:21.31 | Qwell | *somebody* there didn't know who I was, so they issued a chargeback on the card that was used for the room |
| 02:21.45 | cp5 | wait for what? no i didn't hear this |
| 02:21.48 | Qwell | heh |
| 02:22.11 | Qwell | Kerry had me come out for a training course at ITEXPO. Paid for flight+hotel. |
| 02:22.19 | cp5 | oh i didn't know |
| 02:22.21 | Qwell | Pre-paid the room on somebodys card, and apparently didn't tell them |
| 02:22.22 | cp5 | wow that's terrible |
| 02:22.31 | Qwell | so the hotel hit me with it like a month later. |
| 02:22.40 | Qwell | it eventually got all fixed up, but... |
| 02:22.42 | *** join/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis) |
| 02:22.44 | cp5 | i wish you had gotten in touch with me |
| 02:22.46 | cp5 | ahh |
| 02:23.01 | Qwell | I was dealing with Kerry and what's her name...Joyce |
| 02:23.31 | cp5 | ahh |
| 02:29.20 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 02:31.20 | johnakabean | thanks for the comment qwell but where's your degree |
| 02:31.27 | johnakabean | good night |
| 02:31.29 | Qwell | don't have one. |
| 02:31.36 | johnakabean | exactly |
| 02:31.43 | Qwell | okay |
| 02:37.03 | Juggie | hah |
| 02:37.04 | Juggie | nice |
| 02:37.19 | Juggie | clearly the only way to know anything is to get a degree Qwell |
| 02:37.21 | Juggie | didnt you realize |
| 02:37.29 | Qwell | Juggie: yeah.. |
| 02:37.50 | Qwell | cp5: You don't have any formal education like that, right? |
| 02:37.57 | Juggie | school is bullshit |
| 02:38.03 | Juggie | school basically teaches you how to learn |
| 02:38.13 | Qwell | Juggie: I always hated school, heh |
| 02:38.17 | cp5 | Qwell, no, dropped out of HS |
| 02:38.18 | Juggie | there are some areas where school is necessairy |
| 02:38.22 | Juggie | eg, to be a doctor |
| 02:38.24 | Qwell | cp5: highfive |
| 02:38.35 | cp5 | but i'm not claiming to know anything ;) |
| 02:38.40 | Qwell | oh please |
| 02:38.40 | Juggie | but for computers, the bull you learn in school is just a base for what you'll learn on the job |
| 02:38.45 | *** join/#asterisk Nasra (n=maxshipp@CPE001217b1920e-CM00159a010eda.cpe.net.cable.rogers.com) |
| 02:38.59 | Qwell | I know *tons* of people who never did anything like that, and are really smart people or know a ton of stuff regardless |
| 02:39.20 | Juggie | i think thats what i said :P |
| 02:39.22 | Qwell | Juggie: the doctor thing...well |
| 02:39.34 | Juggie | school has its place, but its hardly necessairy to be an expert except in some fields. :) |
| 02:39.38 | Qwell | while I agree with that, I'm not sure it's because you *can't* learn it all on your own |
| 02:39.40 | Juggie | you woudnt want a self tought doctor :P |
| 02:39.47 | Juggie | but you would like a self tought programmer :) |
| 02:39.48 | Qwell | that's my point |
| 02:39.57 | Qwell | there's a stigma with certain things like that |
| 02:40.28 | Qwell | I'm sure that if a self-taught doctor had the same resources available, they'd be just as good as any other doctor |
| 02:40.33 | Qwell | I wouldn't go to one though. heh |
| 02:41.04 | Qwell | a lot of people have that same idea about self-taught computer people though. no idea why |
| 02:41.27 | Juggie | the shit i learned in school was nothing |
| 02:41.28 | Qwell | I didn't really have a point, I guess |
| 02:41.32 | Juggie | :) |
| 02:41.49 | cp5 | "don't worry. i learned this on youtube. i'm making the incision half a millimeter below the rectum" |
| 02:41.56 | Qwell | cp5: totally |
| 02:43.08 | *** join/#asterisk petchaw (n=petchaw@c-66-229-56-147.hsd1.fl.comcast.net) |
| 02:46.24 | petchaw | hello |
| 02:48.42 | cp5 | hello |
| 02:49.39 | petchaw | i am having a hard time enabling the cdr on my server |
| 02:49.51 | petchaw | dont know if anybody did it and might helpo me with it |
| 02:50.44 | petchaw | i wanted to use cdr_odbc |
| 02:54.32 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 02:54.32 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:57.32 | petchaw | can anybody help me configure cdr with cdr_odbc? |
| 03:00.09 | *** join/#asterisk CwizUser (n=tarik@88.244.112.131) |
| 03:00.11 | CwizUser | hello |
| 03:00.18 | *** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman) |
| 03:00.35 | petchaw | hi |
| 03:01.01 | CwizUser | how to use QueuePauseCategory |
| 03:01.08 | CwizUser | astman proxy ? |
| 03:01.21 | CwizUser | Action:QueuePauseCategory |
| 03:01.21 | CwizUser | ??? |
| 03:04.26 | CwizUser | all user is sleeping :((( |
| 03:05.43 | petchaw | i believe they all are |
| 03:05.53 | petchaw | which i could help you, but never used that |
| 03:06.17 | petchaw | me i came here to get some help configuring cdr_odbc to get my cdrs |
| 03:06.26 | CwizUser | $cwiz = new ProgezSantral ( ); |
| 03:06.26 | CwizUser | $cwiz->baglan (); |
| 03:06.26 | CwizUser | $cwiz->komut ( "Action: AgentCallbackLogin\r\nAgent: $dahili\r\nExten: $dahili\r\nContext: extensions\r\n\r\n" ); |
| 03:06.26 | CwizUser | $cwiz->baglantiyikes (); |
| 03:06.30 | CwizUser | simple command |
| 03:06.38 | CwizUser | this is agent login |
| 03:10.19 | CwizUser | :(((((( |
| 03:10.24 | CwizUser | bye |
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| 04:25.56 | vjr | is this thing on? |
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| 04:29.24 | Khratos | ... goes to sleep |
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| 05:22.29 | *** join/#asterisk jbot (i=ibot@rikers.org) |
| 05:22.29 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.5 (2009/01/23), 1.4.23.1 (2009/01/23), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0.3, dahdi-tools 2.1.0.2 (2008/12/18), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev -=- jbot is back! |
| 05:22.29 | carrar | I had 64 days up connect time! |
| 05:22.30 | carrar | LOST!! |
| 05:22.31 | adrianXXX | russellb : how can i fix that ? |
| 05:22.36 | *** join/#asterisk rajiv (n=rajiv@gentoo/developer/rajiv) |
| 05:22.50 | russellb | if you use debian or ubuntu, you can just install the "build-essential" package and be done :) |
| 05:22.52 | Qwell | carrar: donate to FreeNode! |
| 05:22.54 | Qwell | :D |
| 05:22.55 | drmessano | carrar: I would suggest asking freenode for a refund! |
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| 05:23.00 | drmessano | oh |
| 05:23.03 | carrar | heh |
| 05:23.08 | [TK]D-Fender | Clash! |
| 05:23.10 | adrianXXX | Red Hat Enterprise Linux ES release 4 (Nahant Update 4) |
| 05:23.10 | adrianXXX | Kernel \r on an \m |
| 05:23.12 | carrar | or just run my own freenode server |
| 05:23.19 | Qwell | no, but seriously. FreeNode is awesome, and deserves the support. |
| 05:23.22 | drmessano | carrar: Good luck with that |
| 05:23.34 | adrianXXX | i try yum but is dont work... |
| 05:23.34 | carrar | I've had almost every other irc under the sun |
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| 05:23.42 | russellb | Qwell: is there a build-essential like package on RHEL? |
| 05:23.45 | russellb | or would that be too useful |
| 05:23.48 | carrar | multiple efnets, undernet, a ton mor |
| 05:23.49 | drmessano | I wouldnt run a freenode server.. I like my bandwidth |
| 05:23.56 | Qwell | PSA: FreeNode is currently doing a fundraiser to raise money so they can become a charitable org. http://freenode.net/ for more info. :) |
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| 05:24.04 | Qwell | russellb: HA |
| 05:24.10 | russellb | :) |
| 05:24.13 | Qwell | russellb: I wish. |
| 05:24.16 | russellb | nods |
| 05:24.21 | [TK]D-Fender | Qwell: Is that like asking for our money so they can give it away? |
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| 05:24.23 | Qwell | There's the development-tools group, but...meh |
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| 05:24.31 | Qwell | [TK]D-Fender: I |
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| 05:24.34 | drmessano | Its only SORTA complete |
| 05:24.35 | russellb | adrianXXX: then I don't know. I would look up some guides on installing Asterisk on CentOS. the package list should be the same |
| 05:24.35 | Qwell | [TK]D-Fender: I'm glad you asked. |
| 05:24.41 | Qwell | http://blog.freenode.net/2008/10/fundraising-for-charity-status/ |
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| 05:24.47 | adrianXXX | yum install build-essential |
| 05:24.50 | drmessano | Development-tools + some more crap = base for Asterisk |
| 05:24.51 | adrianXXX | No Match for argument: build-essential |
| 05:24.51 | adrianXXX | Nothing to do |
| 05:24.52 | [TK]D-Fender | Qwell: What do I need intermediaries for! |
| 05:24.55 | drmessano | the "Some more crap" is the issue |
| 05:24.56 | Qwell | adrianXXX: groupinstall |
| 05:25.00 | Qwell | instead of install |
| 05:25.16 | russellb | he did build-essential .. |
| 05:25.17 | *** join/#asterisk lucasb (n=lucasb@s154-5-252-231.bc.hsia.telus.net) |
| 05:25.28 | Qwell | adrianXXX: you could just install AsteriskNOW and be done with it :p |
| 05:25.28 | [TK]D-Fender | missing g++ last I saw |
| 05:25.32 | russellb | w00t |
| 05:25.35 | [TK]D-Fender | ew |
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| 05:25.43 | russellb | [TK]D-Fender: don't be a hater |
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| 05:25.52 | adrianXXX | Qwell : how ? |
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| 05:25.57 | Qwell | asterisknow.org |
| 05:26.00 | Qwell | adrianXXX: get the 1.5 beta |
| 05:26.11 | drmessano | Could be worse.. ever heard of Trixbox? |
| 05:26.16 | russellb | Qwell: is 1.5 ever going to be released, or are we going to pull a google? |
| 05:26.23 | carrar | pay for irc |
| 05:26.24 | carrar | heh |
| 05:26.27 | Qwell | russellb: I have no comment. |
| 05:26.29 | russellb | drmessano: yeah, but we still employ the people that work on *NOW |
| 05:26.29 | russellb | heh |
| 05:26.31 | russellb | :-X |
| 05:26.35 | [TK]D-Fender | adrianXXX: http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation |
| 05:26.36 | Qwell | oh burn |
| 05:26.37 | drmessano | Speaking of which.. I was at the gas station the other day |
| 05:26.44 | drmessano | and I got a full service fill up |
| 05:26.51 | drmessano | KERRY GARRISON WASHED MY WINDSHIELD |
| 05:27.28 | drmessano | Did a crappy job too.. but stated they were working on a fix |
| 05:28.20 | [TK]D-Fender | drmessano: Naw, he's pumping gas in Newark.... alongside Elvis! |
| 05:28.20 | [TK]D-Fender | russellb: bai bai :( |
| 05:28.20 | drmessano | I told him my windshield was dirty and he called his squeegee skills "beta" |
| 05:28.20 | [TK]D-Fender | adrianXXX: Do read that link, it'll tell you all the packages you need |
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| 05:34.25 | tamseel | is there any body how can help me |
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| 05:34.25 | tamseel | i have a problem in asterisk |
| 05:34.59 | tamseel | i can make outgoing international calls from my asterisk server |
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| 05:39.05 | carrar | good times |
| 05:39.07 | *** join/#asterisk inv_arp (n=junya@b07s03mr.corenetworks.net) [NETSPLIT VICTIM] |
| 05:39.19 | carrar | should a server just dedicated to Asteirsk products |
| 05:39.31 | *** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) |
| 05:39.38 | Qwell | Why? |
| 05:39.45 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) [NETSPLIT VICTIM] |
| 05:39.49 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) [NETSPLIT VICTIM] |
| 05:39.56 | carrar | irc.asterisk.org !! |
| 05:39.57 | Qwell | FreeNode has always been rather good to us. |
| 05:40.00 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 05:40.01 | [TK]D-Fender | carrar: Grammar is English very your good. |
| 05:40.15 | drmessano | ~netsplit |
| 05:40.16 | jbot | methinks netsplit is something that happens when two IRC servers lose their link, thus isolating the users on every side from each other. a normal part of ALL irc networks, despite what some people bitching about larger networks may seem to think, or an orchestra of poips and thwoops, or something which occurs frequently on OPN |
| 05:40.17 | carrar | yeah it's my fault :) |
| 05:40.27 | carrar | I married Japanese and now I speak Engrish |
| 05:40.27 | drmessano | STOP CREATING DISSENT |
| 05:40.39 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 05:40.40 | *** join/#asterisk jks (i=jks@193.189.93.254) |
| 05:40.48 | drmessano | Japanese girls are hot.. those giant eyes and funny mouths |
| 05:40.51 | drmessano | Wait, thats anime |
| 05:40.54 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) [NETSPLIT VICTIM] |
| 05:40.59 | carrar | oh it's the sam |
| 05:41.02 | carrar | e |
| 05:41.06 | *** join/#asterisk asteriskmonkey (n=philip@69.77.169.14) |
| 05:41.06 | *** join/#asterisk Floppe (i=muppet@crew.cluster-lan.org) [NETSPLIT VICTIM] |
| 05:41.16 | *** join/#asterisk twisted (n=twisted@router.asteriasgi.com) |
| 05:41.51 | carrar | I'd host a server just for Asterisk |
| 05:41.57 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 05:42.13 | Qwell | carrar: not interested |
| 05:42.13 | *** join/#asterisk eric256 (n=eric256@c-75-71-188-88.hsd1.co.comcast.net) [NETSPLIT VICTIM] |
| 05:42.15 | *** join/#asterisk stintel (i=stijn@madwifi/support/stintel) [NETSPLIT VICTIM] |
| 05:42.16 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 05:42.17 | *** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) [NETSPLIT VICTIM] |
| 05:42.25 | *** join/#asterisk Pagautas (n=bigman@ns.voip.ktu.lt) [NETSPLIT VICTIM] |
| 05:42.42 | Qwell | heh, it's interesting seeing some of these hostmasks come in |
| 05:42.43 | drmessano | Freenode is better idea |
| 05:42.46 | *** join/#asterisk jeff (i=jeff@unaffiliated/jeff) |
| 05:42.55 | drmessano | Dont need another server to connect to |
| 05:43.06 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) [NETSPLIT VICTIM] |
| 05:43.11 | Qwell | drmessano: like...yours. |
| 05:43.15 | Qwell | good boy. |
| 05:43.20 | drmessano | heh |
| 05:43.23 | *** join/#asterisk digitalirony (i=digitali@my.grandma.uses.shellium.org) |
| 05:43.34 | *** join/#asterisk hi365_m (n=hi365@85.130.230.240) |
| 05:43.34 | drmessano | I had been running 2 IRC networks |
| 05:43.40 | drmessano | Had to ditch one |
| 05:43.43 | drmessano | But moved it to XMPP |
| 05:43.47 | carrar | I stopped running irc servers in the 90's |
| 05:44.06 | Qwell | Running an IRC network is...hard. |
| 05:44.13 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 05:44.14 | Qwell | of any decent size, anyways |
| 05:44.19 | drmessano | Now, an asterisk XMPP conference would be nice |
| 05:44.22 | Qwell | seanbright: It's not what you think. |
| 05:44.35 | drmessano | IRC is pretty unmanagable.. not friendly at all |
| 05:44.53 | *** join/#asterisk [netman] (n=netman@57.Red-83-63-247.staticIP.rima-tde.net) [NETSPLIT VICTIM] |
| 05:45.00 | drmessano | Services are a hack, but are a single point of failure |
| 05:45.07 | drmessano | but/and |
| 05:45.15 | carrar | Could just run static channels |
| 05:45.19 | Qwell | I like services |
| 05:45.20 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 05:45.27 | Qwell | makes things easy |
| 05:45.33 | drmessano | Services are better than nothing at all |
| 05:45.35 | carrar | and services server if need be |
| 05:45.42 | *** join/#asterisk kisu (n=kisu@2001:5c0:1100:9900:1d37:23b5:94db:ec35) |
| 05:45.45 | drmessano | But IRC is just unmanagable |
| 05:45.57 | *** join/#asterisk fors1 (n=forsen@pat-tdc.opera.com) |
| 05:46.05 | [TK]D-Fender | We seem to have managed for a long time now |
| 05:46.14 | drmessano | Not really |
| 05:46.40 | [TK]D-Fender | drmessano: Leave your psychotic episodes out of this! :p |
| 05:46.52 | *** join/#asterisk stevetotaro (n=Steve@pool-71-254-231-87.hrbgpa.east.verizon.net) |
| 05:48.16 | carrar | 1st bad slit in a few months |
| 05:48.23 | carrar | I'd say it's doing good |
| 05:49.50 | *** join/#asterisk typename (n=chatzill@cpe-68-173-67-220.nyc.res.rr.com) |
| 05:50.38 | drmessano | Um ok |
| 05:51.29 | [TK]D-Fender | carrar: I'd say is WAS going good.. right up until NOW :) |
| 05:51.46 | [TK]D-Fender | conjugates |
| 05:52.21 | carrar | get some of that bailout money for freenode |
| 05:52.27 | Qwell | Too big to netsplit. |
| 05:52.51 | carrar | 50 billion should cover it |
| 05:52.55 | drmessano | IRC is just inherently flawed.. It's incredibly well designed for what it does, but very poorly designed to handle the attacks it's subjected to, and in a security sense. |
| 05:53.27 | *** join/#asterisk zafar_ (n=IceChat7@116.71.208.231) |
| 05:53.34 | [TK]D-Fender | drmessano: That is uncalled for! This is NOTHING like Qwell's chan_skinny botnet! |
| 05:53.49 | drmessano | lol |
| 05:53.52 | carrar | IRC servers should be linked by internal interfaces on a IANA network with dedicated point to point circuits between irc servers ;) |
| 05:53.54 | coppice | IRC's success stems from its great simplicity |
| 05:53.56 | [TK]D-Fender | ;) |
| 05:54.04 | drmessano | coppice: Indeed |
| 05:54.36 | zafar_ | which file should i look for outbound routes |
| 05:54.51 | [TK]D-Fender | zafar_: extensions.conf |
| 05:54.59 | drmessano | With a lot of the advances in IRCd's, it's remarkably self-healing and handled the ebb and flow well.. just has almost no resistance to that same ebb and flow. |
| 05:55.09 | drmessano | handles* |
| 05:55.24 | [TK]D-Fender | drmessano: Ride the wave or get washed away... |
| 05:55.46 | zafar_ | i can see all the extensions there, what pattron should i look for |
| 05:55.51 | drmessano | Riding on a wave chicane? |
| 05:56.19 | [TK]D-Fender | zafar_: Look for? this is YOUR dialplan, go made whatever you WANT |
| 05:56.34 | *** join/#asterisk nix8n82 (n=nate@63.162.27.243) |
| 05:59.23 | *** join/#asterisk dmz (n=dmz@adsl-75-61-102-11.dsl.pltn13.sbcglobal.net) |
| 06:03.00 | *** join/#asterisk dssman (n=no@CPE001d7e602900-CM0011aec52a9c.cpe.net.cable.rogers.com) |
| 06:03.07 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 06:03.07 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:06.17 | drmessano | Anyone know much about ejabberd? |
| 06:08.44 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 06:08.44 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:10.39 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 06:12.11 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 06:13.46 | *** join/#asterisk Chris-NB (n=chris@85-126-61-10.work.xdsl-line.inode.at) |
| 06:15.08 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 06:25.31 | *** join/#asterisk unpaidbill (n=bill@alteredbeastiality.org) |
| 06:26.44 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 06:27.55 | *** join/#asterisk bminish (n=bminish@2001:770:180:0:219:d1ff:fe80:ea64) [NETSPLIT VICTIM] |
| 06:28.00 | *** join/#asterisk viq (n=viq@unaffiliated/viq) [NETSPLIT VICTIM] |
| 06:29.17 | *** join/#asterisk kerx (n=kerx@adsl-69-104-66-38.dsl.irvnca.pacbell.net) |
| 06:29.57 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 06:33.04 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 06:33.07 | *** join/#asterisk fiddur (i=fiddur@c042.rit.se) |
| 06:50.30 | *** join/#asterisk frogonwheels (n=michaelg@203.59.141.93) |
| 06:50.53 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 06:51.10 | zafar_ | can anyone help me by explain this to me "exten => 555,1,Authenticate(1870)" |
| 06:52.13 | frogonwheels | zafar_: Dialing/ calling the context / with 555, will cause a prompt for the password 1870 |
| 06:52.29 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 06:52.52 | frogonwheels | zafar_: show application authenticate |
| 06:52.58 | zafar_ | ah |
| 06:53.14 | zafar_ | thankx buddy |
| 06:53.25 | frogonwheels | np. |
| 06:53.43 | frogonwheels | ~thebook |
| 06:53.44 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 07:01.12 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 07:01.24 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 07:09.38 | *** join/#asterisk _matt (n=matt@mattspc.ipv6.mattstone.net) |
| 07:10.14 | *** join/#asterisk ultrav1olet (n=telnet@94.180.4.213) |
| 07:10.58 | ultrav1olet | What does this message mean? [Jan 27 12:08:37] NOTICE[23096]: chan_iax2.c:9067 socket_process: Rejected connect attempt from 192.168.0.3, request '2567777@incoming' does not exist |
| 07:11.25 | ultrav1olet | I've just set up asterisk with SIP provider and I cannot make any outgoing calls getting this error |
| 07:15.17 | *** join/#asterisk johnakabean (n=none@pool-72-82-113-23.nrflva.east.verizon.net) |
| 07:16.02 | *** join/#asterisk simonr (n=simonr@CPE0018f840e48b-CM0018c0b36b76.cpe.net.cable.rogers.com) |
| 07:16.33 | rajjar | any body tell me about the telecom channel |
| 07:16.52 | rajjar | is there any telecom channel available |
| 07:18.06 | rajjar | help me plzzzzzzzzzz:'( |
| 07:19.03 | *** join/#asterisk CrazyTux (n=brandon@216.138.104.226) |
| 07:19.22 | dssman | @uv, looks like your trying to make an outgoing call on ur inbound context |
| 07:19.32 | dssman | telecom channel? |
| 07:19.33 | *** join/#asterisk Takapa (i=vegard@svanberg.no) |
| 07:19.51 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 07:20.04 | rajjar | yah |
| 07:20.18 | ultrav1olet | dssman: that context is just fine - it defines an extension to call my SIP provider |
| 07:20.37 | dssman | oh, typically inbound would be for incomming calls :P |
| 07:20.37 | rajjar | join romtelecom |
| 07:20.45 | dssman | paste ur config to a pastebin |
| 07:21.27 | *** join/#asterisk PeterFA (n=Peter@unaffiliated/peterfa) |
| 07:21.38 | PeterFA | Anyone know of a free outgoing sip server? |
| 07:22.14 | dssman | termination provider? |
| 07:23.10 | ultrav1olet | dssman: wait a minute |
| 07:23.29 | dssman | k |
| 07:23.39 | dssman | PeterFA -> free to phone line? |
| 07:23.44 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-085afb887b5573ac) |
| 07:24.03 | PeterFA | dssman, I have a VoIP phone and I want to call out for free... know a providor? |
| 07:24.09 | dssman | ohh |
| 07:24.10 | dssman | nope |
| 07:24.34 | dssman | I have heard of a few, for certain areas only tho with a one time fee... they were a little too sketchy for me to look into tho |
| 07:25.06 | *** join/#asterisk dan__t (n=dant@72.233.89.95) |
| 07:25.10 | dan__t | Hello. |
| 07:25.14 | dssman | hey |
| 07:25.25 | drmessano | PeterFA: No such thing as free termination |
| 07:25.44 | dan__t | So, just toying around with a crappy one-line returning AGI script. I'm doing "STREAM FILE beep" |
| 07:25.58 | dan__t | I don't see any problems in the debug or verbose log, both for core and agi, yet I hear no beep. |
| 07:26.13 | dssman | cant help ya there :P |
| 07:26.31 | dan__t | http://pastebin.com/m1c9322db - there's some pastebin love, juts in case. |
| 07:26.34 | dssman | Peter, there are some cheap services out there... you should expect to pay around 1.1c/min |
| 07:26.43 | dan__t | I've toyed around with paths to that beep file, all kinds of good stuff, to no avail. |
| 07:28.23 | dssman | I odnt know AGI... thats next in my adventures... have u tried an absolute path? |
| 07:28.31 | dssman | and are the permissions on the file okay? |
| 07:29.02 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:29.05 | dan__t | The AGI script runs. I can try an absolute path to the beep file... |
| 07:32.28 | dan__t | See, specifying a full path of /var/lib/asterisk/sounds/en/beep.gsm, yields "[Jan 26 23:31:54] WARNING[19632]: file.c:589 ast_openstream_full: File /var/lib/asterisk/sounds/en/beep.gsm does not exist in any format" |
| 07:32.31 | dan__t | Which is expected |
| 07:33.12 | dan__t | I strace'd Asterisk the other day while investigating another problem, and know that it will always try to search ${VARLIBDIR}/asterisk/sounds/{CCODE}, then ${VARLIBDIR}/asterisk/sounds |
| 07:33.22 | dan__t | So, either way, it does find it, and does play it. I just don't hear anything. |
| 07:34.09 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 07:35.52 | dan__t | beh |
| 07:36.21 | dssman | lol |
| 07:37.37 | dan__t | Yea, evne from the AGI manual: Asterisk looks for the file to play in /var/lib/asterisk/sounds |
| 07:40.02 | dan__t | -- Playing 'why-no-answer-mystery' (escape_digits='') (sample_offset 0) |
| 07:40.05 | dan__t | Should work. 100%. |
| 07:40.25 | dan__t | Where's [TK]D to tell me I'm doing something wrong heh |
| 07:41.42 | *** join/#asterisk xrmx__ (n=rm@host1-187-dynamic.31-79-r.retail.telecomitalia.it) |
| 07:42.56 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 07:43.31 | dan__t | Ok, using "SAY DIGITS" works..... wtf. |
| 07:46.51 | dan__t | So does just using a Playback(digits/4), so I know that Asterisk can play GSM files... |
| 07:47.48 | dan__t | Hah. What the F. |
| 07:48.00 | dssman | too tired to play :P |
| 07:50.42 | *** join/#asterisk Gary (n=Gary@freenode/staff/colchester-lug.gary) |
| 07:52.26 | dan__t | Well. I'm out of ideas. |
| 07:57.29 | dssman | Im out of energy |
| 07:57.44 | dssman | every spare minute I seem to play with * |
| 07:58.29 | dssman | u ever play with TAPI? |
| 08:00.16 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 08:00.51 | ultrav1olet | Does anyone know why my asterisk uses extensions.ael only and doesn't want to use extensions.conf? |
| 08:01.28 | dssman | Im out... nite |
| 08:01.31 | dssman | gl ub |
| 08:01.33 | dssman | uv |
| 08:04.10 | *** join/#asterisk maddog01 (n=minotaur@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 08:06.11 | johnakabean | oookay violet, never heard that one |
| 08:06.45 | johnakabean | anyone know why asterisk stopped recording in the mysql CDR.....i have checked cdr_mysql.conf and everything is correct. |
| 08:07.02 | johnakabean | i just recompiled addons |
| 08:09.44 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 08:09.51 | johnakabean | anyone know why asterisk stopped recording in the mysql CDR.....i have checked cdr_mysql.conf and everything is correct. |
| 08:10.07 | johnakabean | i just recompiled addons |
| 08:12.54 | hi365 | maybe you deleted the config file |
| 08:12.56 | hi365 | ? |
| 08:13.27 | johnakabean | have checked cdr_mysql.conf and everything is correct. |
| 08:13.28 | johnakabean | have checked cdr_mysql.conf and everything is correct. |
| 08:13.43 | johnakabean | i even see asterisk connected in mysql manager |
| 08:14.19 | johnakabean | Kill 1100 asteriskuser localhost:46055 asterisk Sleep 79 --- |
| 08:18.24 | fiddur | Hi. I use autopause in queues to have an agent paused when he doesn't answer... I want the agent to be paused in all queues, not just the one he didn't answer. Is there an in-build way for this, or should I make a manager that listens to the pause-event and pauses on other queues... ? |
| 08:27.10 | *** join/#asterisk kadath (n=kadath@rrcs-96-11-226-10.central.biz.rr.com) |
| 08:27.32 | *** join/#asterisk johnakabean (n=none@pool-72-82-106-201.nrflva.east.verizon.net) |
| 08:27.37 | johnakabean | I hate internet explorer 8 |
| 08:28.31 | johnakabean | it was recording the cdr in mysql the whole time but IE 8 just showed calls from yesterday. |
| 08:28.36 | johnakabean | firefox is a different story |
| 08:33.19 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 08:33.59 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 08:34.17 | Zeeek | sex, drugs, rock n roll |
| 08:34.28 | dan__t | found out what my agi problem was, btw. |
| 08:34.37 | dan__t | had to fflish. after. every. fwrite. to. stdout. |
| 08:34.40 | dan__t | That sucks. |
| 08:34.41 | Maliuta | Zeeek: and asterisk |
| 08:35.10 | Zeeek | asterisk schmasterisk |
| 08:35.16 | Zeeek | sex |
| 08:35.20 | dan__t | heh! |
| 08:35.23 | Zeeek | drugs first, depending |
| 08:35.32 | dan__t | rock and roll, always. |
| 08:35.37 | dan__t | Or some variation thereof. |
| 08:35.40 | Zeeek | you need to flush always |
| 08:35.46 | dan__t | Why? |
| 08:35.55 | Zeeek | just the way it is |
| 08:36.05 | dan__t | right. |
| 08:36.19 | dan__t | Well. Alright, I have no problem doubling the number of lines in my code. |
| 08:36.23 | Zeeek | otherwise it acts like a teenager "yeah, yeah, I'll do it when I get time" |
| 08:36.41 | Zeeek | write a function writeandflush() |
| 08:36.42 | dan__t | hah |
| 08:36.56 | dan__t | Uh, suppose I could do that. |
| 08:37.04 | Zeeek | there may be a setting to flush after writes |
| 08:37.14 | Zeeek | it's been a long time since I messed with that |
| 08:37.44 | dan__t | jea |
| 08:38.29 | dan__t | Huh maybe that's what ob_implicit_flush() is for. |
| 08:40.05 | dan__t | Indeed, it is. |
| 08:40.10 | dan__t | implied flush durrr |
| 08:49.53 | dan__t | Ok, let's play a little game of theory. |
| 08:50.18 | dan__t | Say I made this neat dialplan through AGI. Say in that dialplan, there was an option for a caller to be able to call some arbitrary phone number. |
| 08:50.22 | dan__t | What would be the process for that? |
| 08:50.39 | dan__t | Would I use AMI to dial that number, then join it to the current channel? |
| 08:51.16 | lanning | what language? |
| 08:53.08 | dan__t | I'll be doing this in PHP, which I can figure out, I just want to think ahead and know what the process would be within Asterisk. |
| 08:54.01 | lanning | you should be able to turn off buffered IO on a file descriptor/handle |
| 08:54.19 | dan__t | Ohh, that. |
| 08:54.27 | lanning | like $| in perl |
| 08:54.32 | dan__t | Yeah, I've got that sorted. Never written CLI in PHP before. |
| 08:54.43 | dan__t | Hell, never really done anything with Asterisk before, either. |
| 08:54.46 | dan__t | But thank you. |
| 08:55.56 | dan__t | Got any tips for a dialplan like what I had suggested? |
| 08:57.34 | lanning | you mean the play sound? |
| 08:58.29 | dan__t | Got that sorted, too :) |
| 08:58.54 | dan__t | I'm talking about making a phone call in a proxy manner. Some user is on a channel, wants to go ahead and dial some arbitrary number to connect to another party. |
| 08:59.25 | *** join/#asterisk lilalinux (i=e-trolle@fellatio.deswahnsinns.de) |
| 08:59.54 | lanning | you mean like a DISA? |
| 09:00.06 | *** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr) |
| 09:01.20 | lanning | dial into the PBX get a dialtone and be able to dial out? |
| 09:01.37 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 09:01.38 | dan__t | That's a useful one, but what I'd be doing would come after some menus, some authentication etc etc. |
| 09:02.02 | frogonwheels | dan__t: look at DISA |
| 09:02.45 | dan__t | Yeah I just read up on it. My way was a more indirect way. |
| 09:02.53 | dan__t | Authentication would be done prior to any of this. |
| 09:03.07 | lanning | that is just steps in front. |
| 09:03.20 | lanning | you never dial directly into the DISA app. |
| 09:03.22 | frogonwheels | dan__t: ok.. well that's just Authenticate() |
| 09:03.41 | frogonwheels | dan__t: and to do the other bit, you just include=> context_that_dials_out |
| 09:03.41 | lanning | priority 1 answer |
| 09:03.50 | frogonwheels | dan__t: and call Background(dial-now) |
| 09:03.52 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 09:03.56 | frogonwheels | dan__t: or WaitExten(30) |
| 09:04.12 | frogonwheels | dan__t: .. and either loop or whatever. |
| 09:04.17 | *** join/#asterisk mercutioviz (n=chatzill@freeswitch/developer/msc) |
| 09:04.32 | dan__t | hmmm |
| 09:04.50 | frogonwheels | dan__t: it's strangely easy :) |
| 09:04.56 | dan__t | See, I know most all of the pieces, I just need to know how to put them together. |
| 09:05.07 | dan__t | But this puzzle is one of those big pains in the ass, like a 3D puzzle of Big Ben. |
| 09:05.21 | lanning | not really. |
| 09:05.31 | dan__t | Or that 1/3 scale LEGO pneumatic race car. |
| 09:06.03 | frogonwheels | dan__t: http://pastebin.com/d37a5bb43 |
| 09:06.13 | dan__t | I guess my project/goal is to be able to interact with the channel, join another (dialed) extension on that channel, and be able to do things based on DTMF signals, like recording the call, blah blah |
| 09:06.28 | lanning | when you dial to an extension, you run down a list of priorities (like line numbers in BASIC, except you can't skip numbers in your listing) |
| 09:06.34 | frogonwheels | dan__t: there's an autoattend menu with timeouts, multiple choices, and an include => |
| 09:06.35 | dan__t | So in my mind, it's simply a channel, and I manipulate the channel, not send the channel somewhere else. |
| 09:06.46 | dan__t | Yeah, I'm familiar with those. |
| 09:06.51 | frogonwheels | dan__t: yes - but it's running in a context. |
| 09:06.57 | dan__t | Yesa. |
| 09:07.00 | dan__t | Yes, rather. |
| 09:07.11 | dan__t | I understand sending the channel to a new context for another application or function or something. |
| 09:07.18 | frogonwheels | dan__t: have a look - it shows how the include => extensions allows you to dial an extension. |
| 09:07.19 | dan__t | I'm all about making this dialplan as atomic as possible |
| 09:07.24 | dan__t | Will do, brb. |
| 09:07.58 | lanning | sees a flash of blinding white light... |
| 09:08.31 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:08.51 | dan__t | Got it. |
| 09:09.03 | dan__t | Just introduce more extensions into the picture as needed, based on an action. |
| 09:09.38 | dan__t | Then I can do things like if DTMF == 44 goto (somemenu/1) or something |
| 09:09.49 | dan__t | that context would be responsible for maybe dialing out, or whatever |
| 09:11.02 | lanning | right, but you don't do VAR == num, you let the extensions be the menu itself. |
| 09:11.39 | dan__t | yeah, just doing some psuedocode |
| 09:11.50 | lanning | ah |
| 09:12.02 | dan__t | I see now how that 'n' priority can come in handy |
| 09:12.07 | dan__t | Having to inject switches and things like that |
| 09:12.20 | dan__t | I couldn't have imagined having to hack it up without that. |
| 09:12.22 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 09:12.43 | dan__t | I made a little authentication thing based off of raw SQL lookups for SugarCRM. |
| 09:13.11 | dan__t | So, callerid is taken, matched against a number stored, if its there then you're prompted for DTMF input, if you pass... well, haven't gotten that far yet. But still. |
| 09:15.08 | dan__t | Anyway, I'm liking this, it makes my brain hurt. Bad. |
| 09:17.40 | dan__t | Thanks for the help, lanning. |
| 09:17.50 | lanning | np |
| 09:19.17 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 09:21.07 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 09:24.45 | *** join/#asterisk hads (n=hads@argon.nice.net.nz) |
| 09:25.03 | mercutioviz | out of curiousity, does * have T.38 gateway support? |
| 09:28.31 | *** join/#asterisk strummula (n=asdf@static-217-133-202-46.clienti.tiscali.it) |
| 09:29.43 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 09:30.21 | *** join/#asterisk strummula (n=asdf@static-217-133-202-46.clienti.tiscali.it) |
| 09:30.24 | strummula | hello, can you suggest me a sip client? |
| 09:30.28 | strummula | for windows |
| 09:35.41 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 09:37.06 | *** join/#asterisk rvhi (n=chatzill@udp102686uds.hawaiiantel.net) |
| 09:37.19 | rvhi | hi, anyone uses PauseQueueMember? |
| 09:37.29 | rvhi | if it fails, it didn't jump to n+101 |
| 09:39.22 | kaldemar | rvhi: all applications don't support that even if you had priorityjumping=yes set. |
| 09:39.59 | *** join/#asterisk araknattack (n=raul@host89-254-static.186-82-b.business.telecomitalia.it) |
| 09:40.21 | araknattack | say hi |
| 09:40.24 | rvhi | kaldemar: how do i jump then? it says that there is an option |
| 09:40.32 | araknattack | i installed asterisk, how should i configure it? |
| 09:40.43 | rvhi | The option string may contain zero or more of the following characters: |
| 09:40.45 | rvhi | <PROTECTED> |
| 09:42.29 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.com) |
| 09:45.07 | kaldemar | rvhi: what version are you using? |
| 09:45.27 | rvhi | 1.4 |
| 09:45.29 | kaldemar | ~book |
| 09:45.43 | jbot | methinks book is probably 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://tfot.leifmadsen.com or see ~buybook |
| 09:46.33 | kaldemar | araknattack: decide what you want to do with your asterisk, read that book so you start to understand it and then please ask specific questions. :) |
| 09:51.21 | *** join/#asterisk beherit (n=albert@Mihr-RTR03-LF.amdatex.net) |
| 09:51.39 | kaldemar | rvhi: hmm. looking at the code, it should try to jump if the given interface is not found. |
| 09:52.13 | rvhi | kaldemar: i looked at the code too, can't figure out why it didn't jump |
| 09:52.54 | kaldemar | how exactly are you calling the app? |
| 09:53.30 | strummula | sorry for repeat: i'm searching for a windows client that can interface with asterisk. can you suggest one? |
| 09:53.51 | rvhi | PauseQueueMember(queue-1|SIP/101|j) |
| 09:54.17 | *** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 09:56.12 | kaldemar | hm. you could always just check PQMSTATUS in the next priority and do a jump then if necessary. the jumping option is removed in 1.6 anyway. |
| 09:56.53 | araknattack | kaldemar: i just wanted to access the asterisk gui |
| 09:56.55 | kaldemar | and is it so that you don't have SIP/101 defined in queue queue-1? |
| 09:56.59 | araknattack | is that too much to tell? |
| 09:57.08 | kaldemar | araknattack: did you install it? |
| 09:57.11 | araknattack | on the book ther is nothing about it |
| 09:57.14 | araknattack | yes |
| 09:57.23 | araknattack | and i can do asterisk -r |
| 09:57.35 | kaldemar | it's not too much to tell but you didn't say a word about the gui. |
| 09:58.06 | araknattack | well i tought it was the easiest way to configure it |
| 09:58.21 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 09:59.11 | kaldemar | araknattack: the best way is to edit the configuration files directly. |
| 10:01.04 | kaldemar | araknattack: in manager.conf, you have to have webenabled=yes and a defined user, and enable the web interface in http.conf. then restart asterisk. |
| 10:02.45 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 10:05.47 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 10:05.48 | *** join/#asterisk yang (i=yang@CAcert/Assurer/pdpc.supporter.base.yang) |
| 10:07.09 | *** join/#asterisk Faustov (i=user@gentoo/user/faustov) |
| 10:09.04 | araknattack | tnx kaldemar |
| 10:09.28 | Faustov | hello |
| 10:09.48 | Faustov | basically to connect 5 asterisk servers, each server has to register with the other 4, right? |
| 10:10.23 | *** join/#asterisk [Jasper] (n=jverberk@195-240-174-59.ip.telfort.nl) |
| 10:10.25 | [Jasper] | hej guys |
| 10:12.00 | kaldemar | Faustov: the registration is basically just a way to tell the other end where you are, so it is not necessarily needed. but you have to configure clients for all other servers in each server. |
| 10:13.29 | Faustov | kaldemar: thanks, i just found that out from here: http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers |
| 10:14.05 | Faustov | however it doesn't show a good example of how 2 asterisk servers should cooperate based on sip only - what entries in sip.conf would be required? |
| 10:15.27 | Faustov | all my servers got static ip address, so no registering is required, but i suppose i need to implement some authentication and define encryption or digest? |
| 10:17.34 | *** join/#asterisk Dovid (n=annon@tony09-118-62.inter.net.il) |
| 10:18.04 | kaldemar | Faustov: the book has an example on that starting from page 101. |
| 10:18.21 | kaldemar | ~thebook |
| 10:18.21 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 10:18.27 | *** join/#asterisk scurb (n=scurb@194.218.238.2) |
| 10:18.46 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 10:18.55 | kaldemar | "Connecting Two Asterisk Boxes Together via SIP" |
| 10:19.24 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 10:19.53 | Faustov | cool |
| 10:19.55 | Faustov | purchasing |
| 10:20.15 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 10:22.41 | lanning | faustov, it's free, unless you want a dead tree. |
| 10:23.39 | *** join/#asterisk bn43 (n=dhashen@196.212.81.58) |
| 10:24.02 | Faustov | i want healthy eyes, and reading electronic stuff is killing my eyes obviously |
| 10:24.51 | bn43 | hi I've just set up asterisk and can make and receive calls locally - I'm now testing the mailbox feature and find the default voice prompts very scratchy and unclear - yet voice to voice is very good - how can I fix this? |
| 10:26.21 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 10:30.14 | kaldemar | bn43: do you have the gsm coded versions of the sound files? |
| 10:31.32 | bn43 | um how do I check this? I compiled asterisk from source on my ubuntu box and have not changed defaults |
| 10:32.03 | *** join/#asterisk Rabenklaue (n=Rabe@g229218019.adsl.alicedsl.de) |
| 10:32.49 | kaldemar | check whether the files in /var/lib/asterisk/sounds have .gsm extensions. there's something with the gsm codec and gcc >= 4.2. as a workaround you can use sound files in other formats. |
| 10:33.43 | bn43 | yes there are gsm extensions - lots |
| 10:34.21 | bn43 | in fact there are no other type of file extensions |
| 10:34.31 | kaldemar | well, remove *.gsm and take one of these: http://downloads.digium.com/pub/telephony/sounds/ |
| 10:34.32 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 10:34.34 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 10:35.26 | bn43 | which do you recommend? |
| 10:35.50 | kaldemar | wav or alaw/ulaw depending on what codecs you use. |
| 10:35.50 | bn43 | and is there some setting to change or do I just copy the sound files into the sound directory? |
| 10:36.03 | bn43 | using alaw at the moment |
| 10:37.07 | kaldemar | just copy the sound files into the directory. alaw would be nice then to avoid transcoding. |
| 10:37.54 | araknattack | just a question, i need to make simple call recording. |
| 10:38.05 | araknattack | what card do you suggest? |
| 10:38.34 | bn43 | what is transcoding? |
| 10:39.33 | *** join/#asterisk Rabenklaue (n=Rabe@f048080048.adsl.alicedsl.de) |
| 10:41.00 | kaldemar | bn43: transcoding is coding audio from one format to another. it eats resources. |
| 10:41.46 | kaldemar | araknattack: you don't need a card to record calls. |
| 10:42.12 | bn43 | ahh - ok - what is the recommended coding to start with? I'm just testing on an internal network at the moment |
| 10:43.42 | kaldemar | depends on your environment. if you use alaw as your call codec, then alaw is fine. |
| 10:44.48 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
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| 10:48.44 | *** join/#asterisk Gh0sty (n=ghosty@ip-81-11-177-246.dsl.scarlet.be) |
| 10:51.40 | Gh0sty | hello i have a small configuration problem with call queues, |
| 10:51.56 | Gh0sty | http://pastebin.ca/1319423 this is part of the configuration |
| 10:52.09 | Gh0sty | now when i dial 9200 i get immediatly to the SIP/9000 |
| 10:52.23 | Gh0sty | if i dial 9150 i get immediatly the voicemail for 9150 |
| 10:52.40 | Gh0sty | so it seems it does nothing with the queue |
| 10:55.57 | [Jasper] | loader.c:371 load_dynamic_module: Error loading module 'app_dahdibarge.so' |
| 10:56.01 | [Jasper] | what can I do about that? |
| 10:59.34 | *** join/#asterisk MindTheGap (n=MindTheG@201.80.82.57) |
| 11:00.40 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 11:01.39 | fiddur | When app_queue sets Auto-Pause on a realtime queue member, it sends a manager event QueueMemberPaused, but it doesn't say if it was done because the member didn't pick up, or if the interface was busy! I don't want to auto-pause members who are on the phone, just those who doesn't answer whithin timeout! How can I detect this? |
| 11:01.48 | fiddur | do I have to page the interface myself? |
| 11:07.53 | bn43 | kaldemar: thank you very much - the sound is much better now! |
| 11:10.16 | bn43 | just a question - was searching on the forums for this and nothing was coming up - what terms should I have been search for? |
| 11:13.21 | fiddur | ExtensionState (from Manager) returns the same no matter if the phone is occupied or not.. and status is 0 ... the extension is specified from users.conf and is correctly identified in managers answer by Hint that has correct interface... |
| 11:13.48 | fiddur | is there a better way to see if an interface is busy? |
| 11:13.53 | fiddur | (a sip interface) |
| 11:14.07 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:16.01 | bn43 | fiddur: learnt a trick yesterday - u can use the asterisk console to monitor for errors |
| 11:23.38 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 11:36.04 | fiddur | bn43: There is no error message on the console nor the debug-log when my manager issues ExtensionState... |
| 11:36.50 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 11:41.43 | ultrav1olet | How can I create an extension for any unknown (not registered in any extension) number? |
| 11:42.42 | NoxIn- | ultrav1olet: it think somethink like exten => _X. |
| 11:42.55 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 11:43.41 | ultrav1olet | NoxIn-: can it possibly intersect with existing extensions? I mean I have a valid extension and _X. one - will asterisk still work correctly? |
| 11:44.15 | *** join/#asterisk [gnubie] (n=[gnubie]@cm248.omega113.maxonline.com.sg) |
| 11:44.47 | [gnubie] | anybody here uses package management to install asterisk? |
| 11:46.54 | NoxIn- | ultrav1olet: if you have like exten => 1243 and exten => _X. when you dial 1243 it won't take the _X. |
| 11:47.08 | ultrav1olet | NoxIn-: excellent, thank you |
| 11:47.15 | NoxIn- | if they have the same priority |
| 11:48.04 | NoxIn- | now if they have different priority then it will take the one with the first priority |
| 11:48.23 | ultrav1olet | Now camoing back to my yesterdays' question: I cannot link two asterisk servers, one of which runs _only_ as a proxy for another SIP provider |
| 11:49.08 | ultrav1olet | Right now I get this error message from the internal asterisk server: |
| 11:49.11 | ultrav1olet | [2009-01-27 16:33:48] NOTICE[13516]: chan_iax2.c:2991 __auto_congest: Auto-congesting call due to slow response |
| 11:49.11 | ultrav1olet | <PROTECTED> |
| 11:50.38 | ultrav1olet | my internal server has this extension: exten => _9.,1,Dial(iax2/outside_asterisk:password/${EXTEN:1},30,r) |
| 11:50.58 | NoxIn- | <PROTECTED> |
| 11:51.14 | ultrav1olet | roughly so |
| 11:51.33 | ultrav1olet | I get zero messages from B server |
| 11:51.33 | *** join/#asterisk Rabenklaue (n=Rabe@g227165241.adsl.alicedsl.de) |
| 11:51.48 | ultrav1olet | it looks like server B doesn't get any calls from server A |
| 11:57.41 | *** join/#asterisk snafu (n=snafu@p5799EBF6.dip.t-dialin.net) |
| 11:57.46 | snafu | hi everyone :) |
| 12:05.50 | *** join/#asterisk IvanG (n=IvanG@78.52.238.127) |
| 12:06.36 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 12:06.36 | *** mode/#asterisk [+o file] by ChanServ |
| 12:07.06 | [Jasper] | hej guys |
| 12:07.09 | [Jasper] | when is the context used |
| 12:07.11 | [Jasper] | on incomign calls? |
| 12:07.13 | [Jasper] | or outgoing calls? |
| 12:07.15 | [Jasper] | or both? |
| 12:08.07 | frogonwheels | [Jasper]: both |
| 12:08.09 | *** join/#asterisk gambler1 (n=mene_sun@dragan.eunet.yu) |
| 12:09.08 | frogonwheels | [Jasper]: contexts are pretty much used until the call is connected/bridged between two interface / endpoints. |
| 12:09.35 | frogonwheels | [Jasper]: I think my nomenclature could be a bit clearer, but you get the idea. |
| 12:09.40 | *** join/#asterisk Mark17 (n=mark@freenode/sponsor/mark17) |
| 12:09.41 | gambler1 | Does gotoif application supports more conditions? Something like Gotoif device_state = [BUSY or DONTCALL or BLABLA] then jump handlebusy |
| 12:11.06 | kaldemar | ultrav1olet: your dialstring is invalid. you can't use peer:password. if you want to use password in the dialstring, the form is IAX2/local:pass@remote/exten. |
| 12:11.14 | frogonwheels | gambler1: $[ $[${device_state} = BUSY] or . |
| 12:11.17 | *** join/#asterisk prxtien (n=proleone@ppp121-45-69-101.lns10.adl6.internode.on.net) |
| 12:11.20 | kaldemar | ultrav1olet: http://www.voip-info.org/wiki/index.php?page=Asterisk+IAX+channels |
| 12:11.25 | frogonwheels | gambler1: ergh forgotton what the or symbol / thing is. |
| 12:11.34 | prxtien | hey all, whats the best way for me to add the app_transcode module to my 1.6.0.3 install |
| 12:12.34 | frogonwheels | gambler1: $[ $[${device_state} = BUSY] | $[${device_state} = DONTCALL] $] etc... |
| 12:13.32 | ultrav1olet | kaldemar: thanks, wait a minute |
| 12:14.13 | ultrav1olet | kaldemar: what is 'local'? Is it my iax [peer name]? |
| 12:15.19 | gambler1 | frogonwheels: thanks, I thought something like GotoIf($[${DEVICE_STATE(SIP/${EXTEN:1}@myvoipupstreamprovider)} = BUSY|CHANUNAVAIL|CONGESTION|CANCEL|DONTCALL]?:busy) |
| 12:15.34 | kaldemar | ultrav1olet: it is the name of the dialing server, as defined in the other server as peer. |
| 12:15.43 | gambler1 | frogonwheels: but it does not... :) Tnx for help |
| 12:16.57 | snafu | hm |
| 12:17.25 | snafu | anybody knows if the odbc idlecheck patch is avaiable for 1.2.x versions? |
| 12:17.48 | snafu | we're running too much applikations on 1.2.x that we cannot easily migrate to 1.4.x |
| 12:19.04 | *** join/#asterisk path_ (n=path@93-113-21-190.adsl.terra.cl) |
| 12:25.30 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 12:25.32 | *** join/#asterisk HeMan (n=jimmy@193.12.106.19) |
| 12:28.00 | *** join/#asterisk itguru (n=p@host81-134-10-140.in-addr.btopenworld.com) |
| 12:28.52 | itguru | My asterisk server all of a sudden stopped accepting SIP registrations? my entire phone net is down |
| 12:29.17 | itguru | sip show subscriptions comes up blank? |
| 12:32.57 | kaldemar | itguru: subcriptions doesn't list registrations. sip show peers does. |
| 12:33.01 | prxtien | has anyone compiled app_transcode before? |
| 12:34.28 | [Jasper] | hej guys, I'm trying to figure out why asterisk isn't using my context |
| 12:34.33 | [Jasper] | anyone who can help a bit? |
| 12:35.09 | itguru | kaldemar - 0 sip peers? |
| 12:35.18 | kaldemar | [Jasper]: ask a specific question. what context are you referring to? where did you define it? how does it not use it? what are you doing when it doesn't use it? |
| 12:35.21 | itguru | what could cause asterisk to stop accepting connections? |
| 12:37.03 | prxtien | make[1]: *** No rule to make target `gcc', needed by `app_transcoder.so'. Stop. |
| 12:37.04 | prxtien | make: *** [apps] Error 2 |
| 12:37.11 | prxtien | im getting this error when trying to compile my modules |
| 12:38.38 | [Jasper] | kaldemar I made a trunk with a context...I can dial the number...it enters asterisk |
| 12:38.46 | [Jasper] | but goes to default instead of the correct context which I defined |
| 12:39.43 | kaldemar | [Jasper]: pastebin configs and a cli output of a call |
| 12:40.34 | *** join/#asterisk adnc (n=adnc@unaffiliated/adnc) |
| 12:40.58 | [gnubie] | how can i uninstall asterisk-1.4.23.1? |
| 12:41.05 | itguru | Has anyone ever encountered an asterisk box all of a sudden not accepting any connections? How can I test if this is the case? |
| 12:41.34 | adnc | in order to be reachable via your email adresses, do i have to have a registrar server running? |
| 12:41.35 | kaldemar | [gnubie]: how did you install it? |
| 12:41.49 | [gnubie] | kaldemar: make install |
| 12:43.00 | kaldemar | [gnubie]: make uninstall |
| 12:44.03 | *** part/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 12:44.56 | *** join/#asterisk bn43 (n=dhashen@196.212.81.58) |
| 12:45.03 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 12:46.08 | [gnubie] | kaldemar: ok, thanks.. |
| 12:47.10 | [Jasper] | kaldemar <- debug of a call http://www.stringed-up.nl/call.txt |
| 12:47.49 | *** join/#asterisk tamseel (n=IceChat7@116.71.221.106) |
| 12:47.59 | tamseel | hi gyz |
| 12:48.04 | tamseel | i have a problem |
| 12:48.44 | tamseel | i want to send voice mail to my email address i have mentiond to that how can it be possible |
| 12:51.37 | bn43 | hi - I am seeing these errors coming thru from the asterisk console - http://pastebin.com/d7d0e0a67 |
| 12:52.04 | bn43 | everything is working hunky dory otherwise so I'm not sure why this is happening |
| 12:52.17 | [Jasper] | kaldemar seeing anything weird? |
| 12:53.06 | kaldemar | [Jasper]: none of your defined peers match to whoever is calling you, and the one who is calling you is not sending any number, but "s". you'll debug that after you get your peers right. |
| 12:53.38 | [gnubie] | is wondering if there is an asterisk 1.4.23.1 binary rpm for centos-5.2 or binary deb for debian etch or ubuntu 8.04.2 lts |
| 12:54.18 | [Jasper] | hmm |
| 12:54.25 | beek | tamseel: http://tinyurl.com/byd84j |
| 12:54.27 | [Jasper] | kaldemar how will I know whos'calling me? |
| 12:55.20 | kaldemar | [Jasper]: i sure as hell can't tell you who's calling you. :) |
| 12:55.27 | RypPn | bn43 It might help if you also pastebi zaptel.conf, it seems to be referring to it |
| 12:55.51 | bn43 | will do |
| 12:56.15 | RypPn | I have the sneaky suspicion you haven't edited it after running make samples |
| 12:57.22 | tzafrir_laptop | bn43, that message is generated due to a very ugly hack of asterisk-gui - using the asterisk configuration parser to parse zaptel.conf |
| 12:58.01 | bn43 | would you believe it! there is no zaptel.conf in /etc/asterisk! |
| 12:58.06 | tzafrir_laptop | I think it is completely harmless (besides being a pain and hiding other messages in the flood) |
| 12:58.38 | tzafrir_laptop | bn43, grep '#include zaptel.conf' /etc/asterisk/*.conf |
| 12:59.22 | tzafrir_laptop | [gnubie], why do you need that version specifically? |
| 12:59.22 | bn43 | nothing found |
| 13:00.09 | bn43 | RypPn: were u talking to me? |
| 13:00.42 | RypPn | zaptel.conf is usually in /etc bn43 |
| 13:00.49 | [Jasper] | kaldemar why should I define that? |
| 13:00.54 | [Jasper] | whos calling me |
| 13:00.55 | bn43 | cause I did run make samples but there was no instruction afterwards the make |
| 13:00.55 | [Jasper] | as a peer? |
| 13:01.20 | bn43 | at least none that I could see :-) |
| 13:01.27 | [gnubie] | tzafrir_laptop: i got a running 1.4.21 from debian unstable and i got 2 major problems.. for the past few weeks, i don't have time on my own box and i just decided to upgrade it to the latest probably the problems that i have are fixed already |
| 13:01.42 | bn43 | are yes it is in /etc |
| 13:02.56 | kaldemar | [Jasper]: to be able to control where the call lands in your dialplan, i.e. _in what context it goes_. |
| 13:03.11 | *** join/#asterisk kannan (n=kannan@121.246.242.95) |
| 13:03.12 | bn43 | as far as I can see everything is commented out |
| 13:04.57 | tzafrir_laptop | [gnubie], what problems? Are there open bugs for them? |
| 13:04.58 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 13:05.09 | [gnubie] | tzafrir_laptop: the problems are: [1] the callee from a pstn cannot hear me if i am calling from one of my sip phones (one way audio from sip) and [2] the callee (sip phone) from asterisk-b cannot hear me for a minute or less if i am calling from analog or sip phone connected to asterisk-a while peered both asterisk-a and asterisk-b via iax2 |
| 13:05.11 | tzafrir_laptop | (in Debian) |
| 13:06.00 | tzafrir_laptop | if you want a harmless (does-not-break-system) installation from source, try live_ast |
| 13:06.16 | tzafrir_laptop | http://svn.digium.com/svn/asterisk/trunk/contrib/scripts/live_ast |
| 13:06.29 | *** join/#asterisk arpu (n=arpu@chello080109017021.12.14.vie.surfer.at) |
| 13:06.29 | [gnubie] | tzafrir_laptop: i already posted the first problem on the asterisk-users mailing list but since i was so busy with my work, i wasn't able to give a follow-up to what i started on the mailing list |
| 13:08.02 | tzafrir_laptop | [gnubie], please reportbug . Also specifcaly I'm interested to know if the audio is OK (bidirectional) when recorded in ztmonitor |
| 13:08.24 | [Jasper] | kaldemar what i I wanna let ALL calls go to a certain dialplan? |
| 13:09.01 | [gnubie] | tzafrir_laptop: for the 2nd problem, the communication is good for both sip caller and the sip callee connected via iax2 but in the middle of the conversation, the callee will not hear me for a minute or less.. it is like having a pause on the other side.. |
| 13:10.45 | [Jasper] | and what do you mean by the one who is calling me is sending a s kaldemar ? |
| 13:12.24 | [gnubie] | tzafrir_laptop: the 2nd problem has a setup like this: sip_phone =lan-sip=> asterisk-a =internet-iax2=> asterisk-b =lan-sip=> sip_phone |
| 13:13.43 | [Jasper] | I understand that s means no extension kaldemar ...but is that why it goes to default? |
| 13:13.46 | [gnubie] | tzafrir_laptop: for the 1st problem, the setup is like this: sip_phone ==lan-sip==> asterisk ==pots==> analog_telephone |
| 13:13.52 | [Jasper] | it should still go the trunk it's specified context right? |
| 13:14.56 | *** join/#asterisk Khratos (n=khratos@190.166.103.146) |
| 13:14.56 | Khratos | Good morning! |
| 13:16.31 | [gnubie] | tzafrir_laptop: you want me to report the bug on both problems? |
| 13:16.38 | kaldemar | [Jasper]: if the call doesn't match any peer, it goes to default. if it matches a peer but doesn't match any extension, it falls through to default unless you have autofallthrough=no in extensions.conf. but as i said earlier, the call doesn't match any of your peers, that is your first issue. |
| 13:17.19 | tzafrir_laptop | [gnubie], yes, please |
| 13:17.43 | [gnubie] | tzafrir_laptop: ok, i will |
| 13:17.44 | [Jasper] | kaldemar....:( |
| 13:17.51 | [Jasper] | I don't get it...how can it not match ? :p |
| 13:17.57 | [Jasper] | should I rename it or something? |
| 13:19.47 | *** part/#asterisk Gh0sty (n=ghosty@ip-81-11-177-246.dsl.scarlet.be) |
| 13:20.16 | kaldemar | [Jasper]: you don't have a peer with name 0613442399 or ip where the call in your debug comes from. |
| 13:20.56 | [Jasper] | no true kaldemar |
| 13:21.05 | [Jasper] | but that's my mobile phone which I dialed with |
| 13:21.13 | [Jasper] | how can I make a peer for every number which could be dialing me? |
| 13:21.14 | [Jasper] | thta's weird |
| 13:21.16 | prxtien | is there any wnidows softphones with h.264 support? |
| 13:23.47 | kaldemar | [Jasper]: use a context under [general] and take a look at insecure parameter in sip.conf. |
| 13:26.49 | *** join/#asterisk itguru (n=p@host81-134-10-140.in-addr.btopenworld.com) |
| 13:27.53 | itguru | http://pastebin.com/d5801b59f - I'm getting this from my Mitel SIP phone, 5340 - This is it's registration status - I'm guessing this means broken asterisk instance? |
| 13:28.42 | [Jasper] | under general kaldemar |
| 13:28.45 | [Jasper] | so not under default |
| 13:29.03 | kaldemar | sorry, under default that is. my bad. |
| 13:29.46 | kaldemar | mmm.. i'll take that back. it is under general. |
| 13:31.35 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 13:36.51 | *** join/#asterisk propellerhead (n=yogurt2u@host15.190-30-186.telecom.net.ar) |
| 13:37.05 | prxtien | video softphone anyone? |
| 13:37.16 | *** join/#asterisk kamh (n=qmpelkam@xdsl-1817.wroclaw.dialog.net.pl) |
| 13:37.33 | kamh | hi all |
| 13:38.09 | kamh | do U know if there is a problem with CALLERID() function in ast v1.6???? |
| 13:38.57 | kamh | I set it adn it does not change the called number e.g. from 1111 to 555 |
| 13:39.08 | kamh | CALLERID(num)="5555" |
| 13:39.12 | kamh | is it ok? |
| 13:39.53 | kaldemar | drop the ""'s |
| 13:40.27 | kamh | I have£ |
| 13:40.32 | kamh | exten => 100,1,Set(CALLEDRID(num)=555) |
| 13:40.43 | kamh | exten => 100,2,Dial(SIP/1111) |
| 13:41.00 | kamh | but it does not change 100 to 555 |
| 13:41.09 | *** join/#asterisk ickmund (n=ickmund@ada-bcn-fw01.adamoeurope.com) |
| 13:41.09 | *** join/#asterisk fexy (n=fexy@208.3.217.29) |
| 13:41.32 | fexy | Have any of you chaps attempted to use the real time patch for chan-sccp with chan-sccp-b? |
| 13:41.46 | fexy | I'm sure the patch is totally broken and would even compile |
| 13:41.53 | fexy | but has anyone tried? :) |
| 13:42.00 | fexy | err wouldn't |
| 13:42.45 | kamh | I think that there is a problem with CALLERID() functionin v1.6 |
| 13:44.22 | kaldemar | kahyou have a typo in the function name |
| 13:44.53 | itguru | If this disc on which asterisk resides, is full, can this cause any failures in asterisk? |
| 13:45.00 | *** join/#asterisk DarkRift (n=dark@65.92.250.51) |
| 13:45.05 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:45.13 | beek | itguru: A full disk is never a good thing. |
| 13:45.13 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 13:45.16 | beek | Morning jaytee |
| 13:45.22 | jaytee | morning beek |
| 13:45.58 | beek | jaytee: POTS lines were ported from Level 3 yesterday. |
| 13:46.09 | beek | jaytee: PRIs will move on 2/17 |
| 13:46.23 | [Jasper] | kaldemar I expect you wanted me tro try insecure = invite? |
| 13:46.24 | jaytee | from Level 3 to whom? |
| 13:46.29 | beek | jaytee: support will come from a town 45 minutes away. |
| 13:46.55 | beek | jaytee: The company is called D&E. They're a local and cover this part of the state through the center of the state. I've checked around and they have a good rep. |
| 13:47.45 | Faustov | if i have 2 asterisks with static ip, how do i set passwords for each site without using "register"? just secret=xxx on each side in the appropriate [section]? |
| 13:48.05 | kamh | kaldemar: Yes but only here, in ast I have exten => 100,1,Set(CALLERID(num)=555) |
| 13:48.14 | jaytee | so 45 minutes to a couple hours versus 1 to 2 weeks of fingerpointing, 2 to 3 weeks of hemming and hawwing and then 1 and a half weeks actually trying to work the problem. |
| 13:48.56 | beek | jaytee: You got it. It took a little over six weeks to get that damed problem fixed. I can't even tell you how many hours I wasted in the evenings and weekends |
| 13:50.10 | beek | jaytee: I'm still very appreciative of your suggestion for the loop back. With that successful test I was well-armed to fight the "it's CPE" argument I was being given. |
| 13:50.33 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:50.50 | beek | Actually, they have two offices in the two neighboring towns. One is 35 minutes, the other is 45 minutes. |
| 13:51.05 | jaytee | beek, buy O'Reilly's T-1 Survival Guide. The chapter on troubleshooting alone is worth the cover price |
| 13:51.28 | [Jasper] | kaldemar I have no idea what to do here |
| 13:52.26 | lilalinux | somebody here using T.38 with Sipgate? |
| 13:52.48 | beek | jaytee: Already purchased. I now know more than I ever thought I'd need to about T-1s |
| 13:53.17 | beek | But as the song from The Who says: " I won't be fooled again." |
| 13:54.45 | beek | jaytee: Best part about the new company is that I'm getting two PRIs for less than what I was paying for one. |
| 13:57.33 | *** join/#asterisk De_Mon (i=de_mon@fl-67-77-166-5.dyn.embarqhsd.net) |
| 14:00.06 | *** join/#asterisk jjshoe (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 14:00.46 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:01.34 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 14:01.54 | *** join/#asterisk [intra]lanman (n=Raymond@freeswitch/developer/intralanman) |
| 14:06.44 | *** join/#asterisk [acer]lanman (n=Raymond@va-67-76-163-209.sta.embarqhsd.net) |
| 14:07.00 | Faustov | ok i got 2 asterisk servers to talk to each other, but the voice is only one way. ports 5004, 5060, 10000-20000 are open on iptables on both server, [general] has the same selection of codecs - what else could it be? |
| 14:07.35 | snafu | nat? |
| 14:07.36 | [gnubie] | tzafrir_laptop: i already submitted the first problem to submit@bugs.debian.org |
| 14:07.53 | [gnubie] | tzafrir_laptop: and i Cc you |
| 14:08.08 | *** join/#asterisk pecanha (n=e@189.106.180.239) |
| 14:08.12 | Faustov | no nat, both are on public IPs |
| 14:08.33 | [Jasper] | kaldemar did you leave? |
| 14:08.42 | tzafrir_laptop | [gnubie], Package: should be simply asterisk |
| 14:09.18 | tzafrir_laptop | Did you get a bug number? |
| 14:09.18 | [gnubie] | tzafrir_laptop: ah, sorry.. |
| 14:09.23 | [gnubie] | tzafrir_laptop: nope |
| 14:09.27 | tzafrir_laptop | you can also let reportbug handle that for you |
| 14:09.41 | tzafrir_laptop | Let me know when you get it, and I'll reassign the bug to asterisk |
| 14:11.09 | [gnubie] | tzafrir_laptop: ok. i'm waiting for my ticket number |
| 14:14.25 | *** join/#asterisk prxtien (n=proleone@ppp121-45-69-101.lns10.adl6.internode.on.net) |
| 14:14.32 | prxtien | NOTICE[5187]: rtp.c:1586 ast_rtp_read: Unknown RTP codec 127 received from , anyone familiar with this error |
| 14:16.39 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 14:22.26 | [gnubie] | tzafrir_laptop: i got it already... it's the Bug#513229 |
| 14:22.46 | tzafrir_laptop | ok. it somehow worked well |
| 14:23.19 | [gnubie] | tzafrir_laptop: it was only for the 1st problem.. for the 2nd problem, i will submit it maybe tomorrow.. |
| 14:23.50 | [gnubie] | tzafrir_laptop: but, any plans on creating a src deb package for the asterisk v1.4.23.1 for debian stable/testing? |
| 14:23.54 | itguru | I should not be getting 404 errors from my SIP clients, because I can ping the asterisk box. What else would cause 404 errors, even if the network connectivity is correct? |
| 14:24.12 | tzafrir_laptop | BTW: no need to CC us, as we alterdy get this mail (it's sent to the pkg-voip-maintainers list) |
| 14:25.17 | [gnubie] | tzafrir_laptop: i see.. ok.. ;) |
| 14:26.28 | frogonwheels | gambler1: oh - btw that matching problem - I forgot about regular expressions. use the regexp match. |
| 14:27.23 | *** join/#asterisk CrazyTux (n=brandon@216-110-94-230.static.twtelecom.net) |
| 14:28.24 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:28.30 | frogonwheels | $DEVICE_STATE(..) : (A|B|C) |
| 14:30.31 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 14:31.01 | *** join/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec) |
| 14:36.39 | Faustov | it can't be firewall, i've set default policy accept to all |
| 14:36.50 | Faustov | allow=all on both sides |
| 14:37.02 | Faustov | and still the called person can't hear me, i can hear him tho |
| 14:38.55 | *** join/#asterisk The_Boy_Wonder (n=davidvos@nat/digium/x-2ba03dc7dc57e260) |
| 14:43.03 | tzafrir_laptop | [gnubie], those sound files have complete silence (both ways). Were they recorded at the time of a call? |
| 14:43.22 | [gnubie] | no |
| 14:43.52 | [gnubie] | tzafrir_laptop: i was the one who was speaking at that time |
| 14:44.43 | *** join/#asterisk ur8up (n=ktuttle@216.68.250.18) |
| 14:44.48 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 14:45.04 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 14:45.11 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 14:47.22 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 14:47.27 | *** join/#asterisk moy (n=chatzill@bas1-unionville55-1177733953.dsl.bell.ca) |
| 14:49.21 | tzafrir_laptop | [gnubie], maybe you recorded the wrong channel |
| 14:49.24 | tzafrir_laptop | ? |
| 14:49.32 | [gnubie] | nope |
| 14:49.50 | x86 | hey guys, I can't get the time right on my Polycom phones... it sync's with my NTP servers just fine, but the offset is 6 hours ahead of local time |
| 14:50.14 | [gnubie] | tzafrir_laptop: i mean, based on the command that i executed, was it the right one? |
| 14:50.53 | x86 | my offset should be -06:00, but the polycom provisioning files expect it differently, and I'm not sure if it's in seconds or what, but it's showing -21600 |
| 14:51.08 | ur8up | running asterisknow 1.0.2 server seems to loose the ip address from time to time. I have to restart the server. Is this a known issue |
| 14:51.13 | x86 | how do I convert "-06:00" into a format that the polycom configuration files will like? |
| 14:51.23 | *** join/#asterisk drepan (n=pandre@apcdns2.autopage.co.za) |
| 14:51.37 | x86 | ur8up: we don't support asterisknow here, try #asterisknow |
| 14:51.41 | tzafrir_laptop | [gnubie], the call went through Zap/1 ? |
| 14:52.00 | drepan | which is the better codec to use for voice currently? |
| 14:52.05 | ur8up | ok thanks. is this just for asterisk? what is the differnce? |
| 14:52.10 | [TK]D-Fender | x86: that is correct |
| 14:52.20 | pecanha | hey all, is it possible to pass call transfer parameter to RetryDial()? |
| 14:52.22 | [TK]D-Fender | x86: in terms that it is seconds,a nd the math is right |
| 14:52.30 | *** part/#asterisk drepan (n=pandre@apcdns2.autopage.co.za) |
| 14:52.37 | [gnubie] | tzafrir_laptop: zap/1 is fxs |
| 14:52.49 | [gnubie] | tzafrir_laptop: does it mean, i got a wrong one? |
| 14:52.50 | [TK]D-Fender | pecanha: "core show application retrydial" <--- |
| 14:53.24 | tzafrir_laptop | aparantly . The number in the command ztmonitor is the number of zaptel channel to record |
| 14:53.34 | *** join/#asterisk drepan (n=pandre@apcdns2.autopage.co.za) |
| 14:53.48 | *** join/#asterisk Defraz (n=T0tal@72-24-26-22.cpe.cableone.net) |
| 14:54.00 | x86 | [TK]D-Fender: well it's using GMT time I guess, and not local time, since it's 6 hours ahead of local time (As is GMT) |
| 14:54.09 | x86 | [TK]D-Fender: so it's ignoring my offset? |
| 14:54.12 | [gnubie] | tzafrir_laptop: i see.. so, i will try to call again and execute the ztmonitor command |
| 14:54.25 | [TK]D-Fender | x86: Possible. Would help if you PB'd what you're doing. |
| 14:54.34 | drepan | Sorry got dropped, which is the better codec to use for voice? |
| 14:54.42 | [TK]D-Fender | drepan: G.722 |
| 14:55.31 | x86 | [TK]D-Fender: i'm retarded man... that was only set in one phone's config (a phone we no longer have lol) |
| 14:55.37 | *** join/#asterisk thepacmanfan (n=thepacma@173-22-139-185.client.mchsi.com) |
| 14:55.39 | pecanha | [TK]D-Fender: thanks |
| 14:55.41 | x86 | [TK]D-Fender: setting it now in sip.cfg ;) |
| 14:55.57 | itguru | Registration from '<sip:12345@192.168.10.254>' failed for '192.168.10.167' - No matching peer found - What can cause such an error message to come on suddenly? - As in a fully working system, to a not working system? |
| 14:56.04 | [TK]D-Fender | x86: SMRT :) |
| 14:56.18 | drepan | Okay let me rephrase, which codec will give the best quality to least bandwidth ratio? |
| 14:56.22 | [TK]D-Fender | itguru: Bad auth. Plain and simple |
| 14:56.36 | [TK]D-Fender | drepan: G.729 |
| 14:56.52 | thepacmanfan | i want to set up an extension for users to change the time value of a dial command in another extension. what's the cleanest way to do this? AstDB? |
| 14:57.11 | [TK]D-Fender | thepacmanfan: If you don't need to do a ton of these, yes |
| 14:57.11 | drepan | Thanks that is what I thought |
| 14:57.12 | Faustov | omg i made it |
| 14:57.15 | Faustov | i'd like to report a bug |
| 14:57.21 | *** join/#asterisk angryuser (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 14:57.28 | Faustov | default rtp.conf says 10k-20k ports |
| 14:57.34 | Faustov | in fact it tries 5k-31k |
| 14:57.57 | Faustov | well maybe not a bug but it's misleading |
| 14:58.12 | thepacmanfan | [TK]D-Fender, no i don't. the problem i'm running into right now is how to echo the current value in the database back to the use. does SayDigits support variables pointing to a DB? |
| 14:58.35 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:58.46 | [TK]D-Fender | thepacmanfan: AstDB just puts & gets raw text.... you can use this anywhere in your dialplan apps |
| 14:58.50 | drepan | what could cause this? retrans_pkt: Maximum retries exceeded on transmission |
| 14:58.57 | thepacmanfan | even in Swift()? |
| 14:59.59 | [TK]D-Fender | thepacmanfan: Set(DB(abc/123)=Yo fred you suck) Swift(${DB(abc/123)})} |
| 15:00.08 | [TK]D-Fender | thepacmanfan: Set(DB(abc/123)=Yo fred you suck) Swift(${DB(abc/123)}) |
| 15:00.29 | [TK]D-Fender | thepacmanfan: go read up |
| 15:00.45 | [TK]D-Fender | drepan: Bad network setup or other related failure |
| 15:01.06 | *** join/#asterisk thepacmanfan (n=thepacma@173-22-139-185.client.mchsi.com) |
| 15:01.13 | [TK]D-Fender | hopes his name wasn't "fred" :) |
| 15:01.27 | [TK]D-Fender | thepacmanfan: Comical sample, do not take as spite ;) |
| 15:01.32 | thepacmanfan | whoops. could you repeat what you said to me? stupid webchats are too easy to close. |
| 15:01.42 | [TK]D-Fender | [10:00]<[TK]D-Fender>thepacmanfan: Set(DB(abc/123)=Yo fred you suck) Swift(${DB(abc/123)}) |
| 15:01.58 | [TK]D-Fender | thepacmanfan: Assign a value, reference it "whenever" |
| 15:02.00 | *** join/#asterisk n3hxs (n=HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 15:02.03 | x86 | [TK]D-Fender: hmm, set it in sip.cfg, rebooted the phones, still no dice... I'm wondering if I have to format the filesystems on the phones for the changed files to take effect? |
| 15:02.15 | [TK]D-Fender | thepacmanfan: just remember how dead-simple DB2 is and you're set |
| 15:02.21 | [TK]D-Fender | x86: nope |
| 15:02.31 | thepacmanfan | awesome, thanks! i'll give it a shot. i've used the DB before, but never called it from apps like SayDigits or Swift |
| 15:03.06 | [TK]D-Fender | thepacmanfan: its no different for 1 app than another. Values are values. |
| 15:04.24 | x86 | [TK]D-Fender: i'm PB'ing my sip.cfg now |
| 15:04.44 | x86 | [TK]D-Fender: http://pastebin.ca/1319539 |
| 15:05.11 | *** join/#asterisk Nuitari (n=Nuitari@cybernet.nuitari.net) |
| 15:06.07 | Nuitari | why is sip authentication so frustrating? |
| 15:08.39 | thepacmanfan | [TK]D-Fender, so what command would i use to "capture" user keypresses before using Set to save them in the DB? |
| 15:09.08 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:09.20 | [TK]D-Fender | x86: Sure your NTP server is running on that IP? query it direct to make sure its settings are right as well |
| 15:09.32 | [TK]D-Fender | thepacmanfan: "core show application read" |
| 15:10.05 | x86 | [TK]D-Fender: yeah, the phones query the NTP server just fine |
| 15:10.17 | x86 | [TK]D-Fender: the phones have the right minutes, etc, just the wrong offset |
| 15:10.21 | [TK]D-Fender | x86: and make sure its ANSWER is fine |
| 15:10.40 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 15:11.08 | ruben23 | hi nayone have idea on this..?http://pastebin.com/m25721d9b |
| 15:11.46 | [TK]D-Fender | ruben23: says nothing. "sip show peer [peername]" and enable SIP debug and look at what's actually ahppening. |
| 15:11.47 | *** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-4dca3c70c1caae22) |
| 15:13.10 | *** join/#asterisk dlewis (i=c7340d65@about/security/staff/dlewis) |
| 15:14.54 | Nuitari | why would asterisk keep finding a user instead of the peer? |
| 15:14.59 | Nuitari | even if the peer is set properly |
| 15:15.07 | Nuitari | and the identical peer works on another server |
| 15:16.36 | itguru | [TK]D-Fender - Bad authentication from every handset at the same time, when there has been no changes made to the handsets? |
| 15:17.14 | *** join/#asterisk andy-x_l (i=425c01d1@gateway/web/ajax/mibbit.com/x-5ed523722027561f) |
| 15:18.02 | x86 | [TK]D-Fender: answer seems fine |
| 15:19.19 | *** part/#asterisk drepan (n=pandre@apcdns2.autopage.co.za) |
| 15:20.39 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:21.22 | *** join/#asterisk orcimrepus (n=orcimrep@74-130-123-207.dhcp.insightbb.com) |
| 15:21.23 | x86 | [TK]D-Fender: i know the NTP server is fine because my Linksys phones have no problems |
| 15:21.40 | x86 | [TK]D-Fender: for whatever reason these polycom phones just seem to be ignoring the offset |
| 15:21.46 | *** join/#asterisk Teeli (n=tili@58-27-163-244.wateen.net) |
| 15:22.29 | *** join/#asterisk shazaum (n=shazaum@unaffiliated/shazaum) |
| 15:22.57 | *** join/#asterisk otavio_ (n=otavio@debian/developer/otavio) |
| 15:23.13 | jjshoe | x86 are you setting an offset via dhcp? |
| 15:23.33 | [TK]D-Fender | x86: Not sure what to tell you.. the XML segment lokos OK to me... I'd only wonder if its pulling the file ro not... |
| 15:23.42 | otavio_ | Hello ... I'm using asterisk 1.4 with zaptel ... callerid is not working ... do someone has any hint how I can make a test to see if I can fix it? |
| 15:25.53 | beek | otavio_: Are you saying the INCOMING callerid is not working? |
| 15:26.59 | *** join/#asterisk rootforce (n=chatzill@office.aircanopy.net) |
| 15:27.54 | thepacmanfan | [TK]D-Fender, here's my context: http://pastebin.com/m2d05c6fd |
| 15:28.30 | thepacmanfan | i've tried several different formatting options, but Swift keeps breaking at the beginning of the DB part in s,1 |
| 15:30.53 | Nuitari | why, when I have host=therightip port=5060 for a friend in sip.conf asterisk can't find the peer when I call it from the right ip |
| 15:31.06 | thepacmanfan | err, i mean at the db part in s,2 |
| 15:31.56 | *** join/#asterisk loather (n=loather@68.105.249.214) |
| 15:34.01 | otavio_ | beek: yes |
| 15:34.23 | beek | otavio_: POTS lines or BRI/PRI? |
| 15:34.59 | otavio_ | beek: POTS? |
| 15:35.06 | beek | otavio_: Analog |
| 15:35.11 | otavio_ | beek: sorry but I'm not used to telephony language |
| 15:35.16 | otavio_ | beek: yes, analog |
| 15:35.45 | beek | otavio_: And you're sure that the telco is providing you callerid? Have you plugged in a phone w/callerid or a callerid box and ensured that you're really getting callerid? |
| 15:36.18 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 15:39.15 | otavio_ | beek: yes. One of the lines are commenting to another PBX (an analog one) thata also provides callerid and it is also not working |
| 15:40.23 | beek | otavio_: Now I'm confused. You have another PBX connected to the PSTN and it is NOT getting callerid? |
| 15:40.52 | *** join/#asterisk RMod (n=nicolasj@unaffiliated/rmod) |
| 15:40.59 | Faustov | yay, made a mesh of 3 asterisk servers |
| 15:41.03 | Faustov | 2 more to go... |
| 15:41.46 | Nuitari | I've got a mesh of 10 servers, they all can talk to each other just fine, except 2 of them, and only one way |
| 15:41.48 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 15:42.07 | Nuitari | and I can't find the problem and I'm about to give up throw it all away and call cisco |
| 15:43.30 | otavio_ | beek: this have three ports |
| 15:43.38 | otavio_ | beek: two, are analog POTS |
| 15:43.54 | otavio_ | beek: and one connect to another PBX (an analog one) |
| 15:44.31 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 15:44.31 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:44.57 | beek | otavio_: I understand. Have you plugged just a phone into the POTS lines to see if you get callerid information from your Telco? |
| 15:45.37 | Faustov | Nuitari: i just had a similar problem, the default rtp ports are different from the ones defined in the default config (was my solution) |
| 15:45.43 | Faustov | but 10... damn :> |
| 15:45.52 | Faustov | ur configs must have 50KB |
| 15:45.52 | Faustov | :> |
| 15:45.57 | Nuitari | in this case I don't even get authentication |
| 15:45.58 | Faustov | if not more |
| 15:46.02 | Nuitari | it just doesn't see it as the peer |
| 15:46.06 | *** join/#asterisk stevetotaro (n=Steve@pool-71-254-231-87.hrbgpa.east.verizon.net) |
| 15:46.15 | Nuitari | and it's only one way, and it's only these 2 servers |
| 15:47.03 | Nuitari | Faustov: the joys of scripts |
| 15:47.15 | otavio_ | beek: the PBX gets the callerid from their extensions but those are not provided |
| 15:48.18 | *** part/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 15:48.23 | beek | otavio_: Lets start from the beginning. When someone calls your number you do not get their callerid information. Is that the problem? |
| 15:48.30 | dlewis | Nuitari: if you throw it away, please let me know... i haven't been dumpster diving in a few years... |
| 15:48.40 | otavio_ | beek: yes ... |
| 15:48.57 | otavio_ | beek: and when someone behind the other PBX calls I also do not get it |
| 15:49.39 | Nuitari | dlewis: with the amounts of frustration that this is causing me I'll smash everything with a sledgehammer first |
| 15:50.01 | *** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com) |
| 15:50.04 | beek | otavio_: Pastebin your /etc/asterisk/zapata.conf file |
| 15:50.05 | prxtien | Nuitari, firewall issue? |
| 15:50.08 | beek | ~pastebin |
| 15:50.09 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:50.13 | dlewis | Nuitari: lol |
| 15:50.34 | Nuitari | prxtien: doubt it, the server can talk to the 9 others fine |
| 15:50.35 | beek | otavio_: You should have a "usercallerid=yes" in that file somewhere. |
| 15:50.43 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-53b548d4e897b2e8) |
| 15:50.43 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:50.54 | prxtien | routing? |
| 15:51.20 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:51.47 | Nuitari | prxtien: no the packets reach the other server, the problem is that it doesn't do the peer authentication |
| 15:51.50 | otavio_ | beek: http://paste.debian.net/27024/ and http://paste.debian.net/27025/ |
| 15:51.52 | prxtien | ah |
| 15:52.01 | Nuitari | it skips it then tries with the user I'm calling from |
| 15:52.02 | prxtien | iax2 trunk? |
| 15:52.10 | Nuitari | sip |
| 15:52.14 | prxtien | oo |
| 15:52.23 | prxtien | how come not iax2 trunks between servers? |
| 15:52.24 | Nuitari | iax2 ties all calls to one cpu |
| 15:52.45 | [TK]D-Fender | thepacmanfan: pastebin your failed attempt |
| 15:52.52 | [TK]D-Fender | thepacmanfan: along with DB dumps from CLI |
| 15:53.12 | prxtien | this app_rtsp module is driving me nuts |
| 15:53.17 | prxtien | ive given up on it with 1.6 |
| 15:53.19 | Nuitari | though I might do iax2 between these 2 and just give up |
| 15:53.20 | *** join/#asterisk CapriCoRN^80 (n=int@207.176.6.160) |
| 15:53.32 | CapriCoRN^80 | hi all |
| 15:54.53 | beek | otavio_: It looks okay for the U.S. |
| 15:55.08 | otavio_ | beek: heh, I'm in Brazil ;-) |
| 15:55.21 | otavio_ | beek: just a little far :P |
| 15:55.38 | otavio_ | beek: but let's solve one problem by time. How I could check the PBX2 issue? |
| 15:55.52 | CapriCoRN^80 | hi [TK]D-Fender |
| 15:56.23 | beek | otavio_: I'm less interested in the PBX2 issue. If we get it fixed for your Telco then it should work for your PBX2 as well. |
| 15:57.03 | beek | otavio_: What version of Asterisk? |
| 15:57.08 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 15:57.52 | otavio_ | beek: 1.4.21 |
| 15:58.16 | beek | otavio_: I don't know the callerid specs for Brazil. Are they the same as in the US? |
| 15:58.41 | Nuitari | well iax2 works |
| 15:58.45 | Nuitari | good enough |
| 15:59.19 | *** join/#asterisk wonderworld (n=ww@ip-62-143-28-129.unitymediagroup.de) |
| 15:59.28 | tzafrir_laptop | otavio_, caller ID for analog (POTS) in Brazil doesn't work for you? |
| 15:59.50 | otavio_ | tzafrir_laptop: exactky |
| 16:00.01 | *** join/#asterisk mnicholson (n=mnichols@nat/digium/x-a1898396ff08248c) |
| 16:00.11 | otavio_ | I'm interested to know if there're a way to test another signaling or something like |
| 16:00.22 | otavio_ | beek: not sure.. |
| 16:00.23 | tzafrir_laptop | http://bugs.digium.com/view.php?id=9096 |
| 16:00.49 | tzafrir_laptop | :-( |
| 16:01.36 | otavio_ | tzafrir_laptop: let me try this |
| 16:01.49 | *** join/#asterisk citats (n=james@mrplow.gnuinternet.com) |
| 16:01.56 | *** join/#asterisk bmg505 (n=leon@196.209.235.43) |
| 16:02.02 | beek | otavio_: My googling points to issues with Brazil & CallerID. Look up what tzafrir_laptop posted. |
| 16:03.13 | otavio_ | wctdm doesn't look to support dtmf=1 |
| 16:04.00 | beek | otavio_: The bug he pointed you to has some patches. |
| 16:04.07 | otavio_ | hummm found the patch |
| 16:05.42 | *** join/#asterisk ming_zym (n=ming_zym@125.39.45.16) |
| 16:10.23 | otavio_ | tzafrir_laptop: I noticed you commited it |
| 16:10.31 | otavio_ | tzafrir_laptop: does Debian zaptel source has it? |
| 16:10.49 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 16:12.01 | tzafrir_laptop | otavio_, there's a pending patch there |
| 16:12.42 | otavio_ | tzafrir_laptop: where? |
| 16:12.42 | otavio_ | tzafrir_laptop: there're many patches :P |
| 16:13.23 | *** join/#asterisk mercutioviz (n=chatzill@freeswitch/developer/msc) |
| 16:13.38 | tzafrir_laptop | that patch is a patch to our specific driver that uses a similar approach |
| 16:14.13 | tzafrir_laptop | and latest patches there are vs. dahdi . They may require some small adjustments to apply to zaptel |
| 16:14.22 | dlewis | tzafrir_laptop: the echo cancellation software offered by digium, will that work for the x100p card? |
| 16:15.09 | tzafrir_laptop | dlewis, hpec? It will work (as for any Zaptel/DAHDI driver) |
| 16:15.11 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-cd72eaa642f8ca4b) |
| 16:15.11 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:15.30 | dlewis | ko |
| 16:15.32 | dlewis | ok |
| 16:15.45 | otavio_ | tzafrir_laptop: and there're dahdi packages for Debian? |
| 16:15.52 | otavio_ | tzafrir_laptop: compatible with lenny? |
| 16:15.59 | tzafrir_laptop | there are. Not yet uploaded |
| 16:16.10 | tzafrir_laptop | as there's a compatibility issue |
| 16:16.17 | otavio_ | tzafrir_laptop: which? |
| 16:16.18 | [TK]D-Fender | dlewis: Give OSLEC a shot firs |
| 16:16.21 | otavio_ | tzafrir_laptop: I can build and test, np |
| 16:16.22 | tzafrir_laptop | And you'll have to have a version of Asterisk that uses DAHDI |
| 16:16.40 | otavio_ | tzafrir_laptop: lenny one doesn't support it? |
| 16:17.03 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 16:17.09 | dlewis | [TK]D-Fender: ok, thanks man |
| 16:17.35 | dlewis | I was just searching for an os version |
| 16:17.48 | tzafrir_laptop | otavio_, dahdi/zaptel is a compile-time option, even if we did use 1.4.22 or later |
| 16:18.31 | otavio_ | tzafrir_laptop: oh .. that would require me to build asterisk ... what i'd like to avoid :P |
| 16:18.58 | itguru | As soon as I dial - even with the output of http://pastebin.com/d748a2f03 - I get a straight engaged tone - which I thought *should* be impossible |
| 16:20.41 | [TK]D-Fender | itguru: umm... what is that supposed to mean? |
| 16:21.33 | itguru | [TK]D-Fender as in it is identical to every other extension that I have, it registers normally, but can't make calls |
| 16:21.52 | *** join/#asterisk sigmounte (n=sigmount@bai59-1-88-172-80-96.fbx.proxad.net) |
| 16:21.55 | itguru | even though another handset, configured exactly the same, can make calls |
| 16:22.25 | [TK]D-Fender | itguru: And you aren't showing us the SIP debug of the failed attempt |
| 16:22.42 | [TK]D-Fender | itguru: Nor device configs (which we equally have no reason to trust) |
| 16:22.47 | sigmounte | hi ! i'm using asterisk 1.6.x and the command MONITOR , but it look like it does record anything , how can i debug it (i'm in -rvvvvvvvvv and does not see anything relatd to monitor ) thanks |
| 16:22.49 | [TK]D-Fender | itguru: Its failing for a reason. |
| 16:23.11 | [TK]D-Fender | sigmounte: Pastebin what you're doing and mayb e we can tell you what's wrong. |
| 16:23.13 | [TK]D-Fender | ~pb |
| 16:23.14 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 16:23.15 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 16:23.55 | sigmounte | [TK]D-Fender, you want a pastebin about what my log in the asterisk console ? or of my command MONITOR in my extension.conf ? |
| 16:24.08 | [TK]D-Fender | sigmounte: The entire call from beginning to end |
| 16:24.30 | *** part/#asterisk ur8up (n=ktuttle@216.68.250.18) |
| 16:25.04 | CapriCoRN^80 | [TK]D-Fender: I have a server which has a private ip address scheme.The System is connected using a wireless router which got public ip address . I am using that server right now as i opened its ssh port. Asterisk is configured on that server and now i want softphone from other networks can use that server |
| 16:25.15 | CapriCoRN^80 | what should i do in that case ? |
| 16:25.32 | [TK]D-Fender | CapriCoRN^80: what "other networks"? |
| 16:25.36 | CapriCoRN^80 | i haave few things in mind that i will open 5060 port in that wireless router which is connect to that server |
| 16:26.15 | itguru | [TK]D-Fender I'm get a sip debug of the failed call |
| 16:26.26 | CapriCoRN^80 | [TK]D-Fender: other networks means that i am in different network now with my laptop and i want that my softphone got working using that server |
| 16:27.39 | [TK]D-Fender | CapriCoRN^80: then you are implying "anywhere out on the internet" |
| 16:27.52 | [TK]D-Fender | CapriCoRN^80: with that in mind , READ UP : |
| 16:27.53 | *** join/#asterisk bmoraca (n=bmoraca@209.60.253.58) |
| 16:27.54 | [TK]D-Fender | ~sipnat |
| 16:27.55 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:27.57 | [TK]D-Fender | ^^^^^^^^^ |
| 16:28.27 | CapriCoRN^80 | [TK]D-Fender: right now i am using that server using ssh. It means that if open port 5060 on that wireless router . i hope that will work |
| 16:28.32 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 16:28.44 | [TK]D-Fender | CapriCoRN^80: you need to do a LOT more. READ THE GUIDE |
| 16:28.50 | itguru | [TK]D-Fender http://pastebin.com/d2f0be006 - This is the SIP debug of the failed call |
| 16:29.07 | rootforce | what is the list of questions that you can ask jbot? |
| 16:29.14 | CapriCoRN^80 | ok |
| 16:29.15 | *** join/#asterisk shido6 (n=shido6@74-132-200-214.dhcp.insightbb.com) |
| 16:29.57 | [TK]D-Fender | itguru: SIP/2.0 488 Not acceptable here |
| 16:30.28 | [TK]D-Fender | rootforce: For things its been trained for like these last few tips, there is no "dump" feature |
| 16:31.08 | sigmounte | [TK]D-Fender, http://pastebin.com/m49bce318 |
| 16:32.19 | [TK]D-Fender | sigmounte: I think DISA is screwing you up. Test without it. |
| 16:32.26 | sigmounte | ok |
| 16:32.47 | itguru | [TK]D-Fender - It's a codec issue? |
| 16:33.35 | [TK]D-Fender | itguru: It refuses before showing the comparative list, and I see ulaw/alaw in common from what you have shown. I'm wordering about enforced encryption |
| 16:33.53 | [TK]D-Fender | a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Y2UwMDg3NzA4MmNmNmZlNmNlMDA4NzcwODJjZjZm|2^20 |
| 16:33.55 | [TK]D-Fender | a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:MzgxM2U3NTkwNzY3YmUyYTM4MTNlNzU5MDc2N2Jl|2^20 |
| 16:33.58 | itguru | wordering..... I'm guessing you mean wondering! |
| 16:34.12 | russellb | wordering is just wondering in words. |
| 16:34.20 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:34.30 | [TK]D-Fender | INDEED |
| 16:34.38 | [TK]D-Fender | itguru: This is my current theory.. |
| 16:35.10 | [TK]D-Fender | itguru: 488 is usually a codec issue, but the pieces look OK. Sometimes its an "OMG you can't report that IP for RTP", but that looks sane as well |
| 16:37.46 | ruben23 | [TK]D-Fender:...? |
| 16:37.52 | *** join/#asterisk keith4 (n=keith@lust.CC.Lehigh.EDU) |
| 16:38.35 | beherit | Hello guys, I have a prevoiusly working * box, when I tried the MySQL Realtime I was not able to call even the local extension. any idea? |
| 16:38.57 | ruben23 | whan i tried to call there is no ring tone or sign of connection and this is my CLI output http://pastebin.com/m25721d9b it say no route destination... |
| 16:39.17 | ruben23 | and also this http://pastebin.com/m1ec2f195 |
| 16:39.58 | [TK]D-Fender | beherit: Idea : pastebin actual call debug for us to look at |
| 16:40.34 | [TK]D-Fender | ruben23: and I told you to enable SIP DEBUG and look at the call |
| 16:40.59 | ruben23 | <PROTECTED> |
| 16:41.38 | beherit | [TK]-Fender- Ok wait |
| 16:43.20 | ruben23 | what this mean===> Jan 28 00:32:10 NOTICE[7830]: rtp.c:587 ast_rtp_read: Unknown RTP codec 126 received it always appear.. |
| 16:43.35 | *** join/#asterisk b0lt (n=b0lt@rh-101-205.greensburg.resnet.pitt.edu) |
| 16:44.21 | *** join/#asterisk beherit (n=albert@netsys.bts.corp.amdatex.net) |
| 16:44.49 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 16:45.01 | rue_mohr | so! there is a new version of dahdi out! |
| 16:45.12 | rue_mohr | which seems to include support for oslec! |
| 16:45.55 | [TK]D-Fender | ruben23: Means "I don't know that codec" |
| 16:45.57 | *** join/#asterisk scud (n=scud@69.73.16.86) |
| 16:46.24 | rue_mohr | or so it seemed last night |
| 16:46.25 | b0lt | hmm |
| 16:46.37 | b0lt | does anyone know if the tigerjet 560C is usable with asterisk? |
| 16:48.10 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:48.19 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:50.34 | rue_mohr | ARG, every time I'm looking for zaptel all I can find as dahdi, and every time I'm looking for dahdi all I can find is zaptel |
| 16:51.34 | pecanha | hey guys, do x-lite support call transfer? |
| 16:51.40 | rue_mohr | http://downloads.digium.com/pub/ <-- cant find an dahdi there |
| 16:51.42 | [TK]D-Fender | b0lt: http://www.earth.li/~noodles/hardware-usbfxs.html |
| 16:51.50 | *** join/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
| 16:52.28 | rue_mohr | found it, whys it hidden under there |
| 16:53.15 | rue_mohr | I'm confused, when I opened dahdi-linux-2.1.0.3.tar.gz last night at home there was a oslec entry in it, but the one I have dons't |
| 16:54.55 | *** join/#asterisk vader-- (n=me@c-71-225-195-86.hsd1.nj.comcast.net) |
| 16:55.43 | rue_mohr | <PROTECTED> |
| 16:56.45 | [gnubie] | tzafrir_laptop: i already attached the right one.. kindly check your inbox.. thanks.. ;) |
| 16:56.48 | b0lt | [TK]D-Fender: thanks |
| 16:57.03 | b0lt | [TK]D-Fender: i can't tell if that does on/off hook detection though |
| 16:57.56 | beherit | [TK]D-Fender: here it is http://pastebin.com/m7898b41b |
| 16:58.48 | tzafrir_laptop | rue_mohr, it's disabled by default |
| 16:58.53 | tzafrir_laptop | in Kbuild |
| 16:59.12 | rue_mohr | ah, please help me or point me to help or something |
| 16:59.12 | [TK]D-Fender | beherit : [Jan 28 01:54:50] VERBOSE[12465] logger.c: Looking for 6001 in from-internal (domain 192.168.10.16) SIP/2.0 404 Not Found^M |
| 16:59.18 | tzafrir_laptop | See the section about oslec in the README: http://docs.tzafrir.org.il/dahdi-linux/#_oslec |
| 16:59.21 | [TK]D-Fender | beherit : Go fix your dialplan |
| 17:01.02 | rue_mohr | tzafrir_laptop, drivers/dahdi/Kbuild ? |
| 17:01.59 | rue_mohr | are there options on make I can use to have oslec support enabled? |
| 17:02.26 | *** join/#asterisk WhiteWolf (i=whitewol@i-am.whitew0lf.info) |
| 17:02.43 | tzafrir_laptop | rue_mohr, yes. But see that section above |
| 17:04.46 | rue_mohr | tzafrir_laptop, I did that source merge thing before and it landed me with a driver that wouldn't load every time I turned on echo canceling |
| 17:05.16 | rue_mohr | in a half hour I'm gonna try going back to the zaptel drivers if I cant get anywhere |
| 17:06.32 | beherit | [TK]-Fender: ok thanks. |
| 17:06.45 | *** join/#asterisk RobH (n=RobH@120-118.186-72.tampabay.res.rr.com) |
| 17:06.50 | rue_mohr | will someone help me work out why the dahdi driver wont load when I turn on echo canceling? |
| 17:07.21 | rue_mohr | no? |
| 17:07.59 | *** join/#asterisk rwaite (n=fieldyca@rrcs-74-218-125-86.central.biz.rr.com) |
| 17:10.18 | rue_mohr | downloads the zaptel drivers |
| 17:10.57 | *** join/#asterisk tobias (n=tobias@cpe-069-134-127-101.nc.res.rr.com) |
| 17:11.43 | *** join/#asterisk macros73 (n=cs@dsl093-063-232.pit1.dsl.speakeasy.net) |
| 17:12.08 | rue_mohr | http://www.freepbx.org/forum/freepbx/installation/asterisk-1-6-dahdi-oslec |
| 17:12.24 | rue_mohr | those were the directions I followed, with fixed directories |
| 17:13.10 | *** join/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc) |
| 17:13.58 | rue_mohr | echocanceller=mg2,5-8 <- it loads with that set... |
| 17:14.36 | jaytee | but that's not oslec, it's mg2 |
| 17:15.15 | rue_mohr | yes and the dahdi loads in asterisk with that set, but as soon as I set it to oslec, the dahdi goes poof, and I have no idea what went wrong |
| 17:15.31 | rue_mohr | or how to find out |
| 17:16.43 | jaytee | and tzafrir posted a link to a doc on his website. did you read it? it means dahdi support for oslec is still in the "experimental" stage. |
| 17:17.12 | rue_mohr | I would think it should atleast load |
| 17:18.22 | rue_mohr | [Jan 27 09:15:55] ERROR[4972] chan_dahdi.c: Unable to register channel '6' <- comes up in messages when i have oslec turned on |
| 17:18.32 | *** join/#asterisk af_ (n=getsmart@88-149-230-97.dynamic.ngi.it) |
| 17:19.12 | rue_mohr | ok, so I either go back to zaptel, which it sounds like oslec works fine with, or I buy the hardware echo can |
| 17:21.30 | *** join/#asterisk ftp3 (n=none@pool-71-117-187-57.ptldor.dsl-w.verizon.net) |
| 17:22.00 | *** join/#asterisk keebler (n=keebler@h1.224.20.98.dynamic.ip.windstream.net) |
| 17:23.04 | [gnubie] | waves.. gtg now.. thanks.. ;) |
| 17:23.22 | jaytee | I'd buy the hardware echo cancellation but then we've beaten this dead horse before, last week in fact IIRC |
| 17:24.34 | rue_mohr | yup, I give up, the office people said their ok if I buy the card |
| 17:24.45 | *** join/#asterisk doger (n=daniel@mail.ipcontact.com.uy) |
| 17:24.54 | rue_mohr | but I'd like it to be known OSLEC DOES _NOT_ WORK WITH DAHDI |
| 17:25.01 | doger | hi, a queue question |
| 17:25.08 | ftp3 | hey.. i am trying to troubleshoot why my did isn't ringing.. can someone tell me the command to watch the asterisk logs in real time so I can see what I am doing wrong? :-) |
| 17:25.42 | doger | is there some way to do a roundrobin strategy in 1.4 ? I mean a call A->B->C->D, always same behaviour |
| 17:26.08 | doger | ftp3, try "tail -f /var/log/name_of_asterisk_log" |
| 17:26.19 | [TK]D-Fender | ftp3: "asterisk -rvvvvvvvvvvvvvvvvvv" |
| 17:26.28 | [TK]D-Fender | ftp3: and then "sip set debug" |
| 17:26.29 | Qwell | doger: there's rrmemory |
| 17:27.05 | [TK]D-Fender | dogAlways starts with "A" Or that a new call resumes where it last left off? |
| 17:28.31 | doger | I want to always starts with "A". I could do this in 1.2 with roundrobin, and putting B penalty 1, C penalty 2, D penalty 3, etc. |
| 17:28.56 | [TK]D-Fender | doger: IIRC the old "roundrobin" is pretty much completely removed as a strategy |
| 17:29.23 | Qwell | [TK]D-Fender: in favor of rrmemory |
| 17:29.27 | doger | i know that roundrobin is deprecated, but I don'k know actually was removed or deprecated... Someone's have a good argument for doing this ? |
| 17:29.32 | *** join/#asterisk rvhi (n=chatzill@udp102686uds.hawaiiantel.net) |
| 17:29.42 | [TK]D-Fender | doger: RRMEMORY was the one in 1.2 that remembered where it left off, and 1.4 absorbed it into "roundrobin" |
| 17:30.02 | [TK]D-Fender | doger: Guess nobody cared about the old style.. |
| 17:30.13 | [TK]D-Fender | doger: I don't see a practical reason to remove a strategy like that |
| 17:30.51 | doger | I think that adding again the "break" statement in app_queue.c for roundrobin I'll get the old roundrobin strategy |
| 17:31.11 | ftp3 | thank you :-) |
| 17:31.35 | *** join/#asterisk otavio (n=otavio@debian/developer/otavio) |
| 17:32.16 | manxpower | In the Asterisk world "deprecated" = "will be removed in the next major release" |
| 17:33.02 | doger | yes I know |
| 17:33.12 | doger | probably I'll ask in asterisk-dev |
| 17:33.19 | *** join/#asterisk bn43 (n=dhashen@41.26.95.203) |
| 17:34.40 | beherit | [TK]D-Fender: thank for pointing out the root cause of my problem its now working |
| 17:34.59 | [TK]D-Fender | doger: You'll want to think twice about mixing thisk 1.2" and "DEVELOPMENT" in the same sentence :p |
| 17:35.21 | [TK]D-Fender | doger: You'll want to think twice about mixing things like "Asterisk 1.2" and "DEVELOPMENT" in the same sentence :p |
| 17:35.33 | jaytee | lol |
| 17:35.44 | beherit | i have another question whats the cause of this error---- " app_dial.c:1286 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)" |
| 17:36.15 | angryuser | beherit it is unable to create the sip channel |
| 17:36.32 | [TK]D-Fender | beherit : LAST TIME : that message is meaningless junk by itself |
| 17:36.46 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 17:36.47 | [TK]D-Fender | beherit :Enable full debug and open your eyes |
| 17:37.43 | bn43 | hello I'm trying to enable recording on my snom320 phone - I'm following this guide http://asterisk.snom.com/index.php/Asterisk_1.4/Call_Recording |
| 17:37.57 | beherit | angryuser: yeah what might cause this problem? |
| 17:38.09 | [TK]D-Fender | beherit : It could be ANYTHING |
| 17:38.56 | bn43 | but I'm confused by the last bit - [my_context] - I used asterisk-gui to start of now when I am looking at the actual extensions.conf, I can't find anything similar to mycontext |
| 17:40.03 | [TK]D-Fender | bn43: Because when the GUI parses your users.conf it generates extens LIVE that aren't part of your extensions.conf into a context all its own |
| 17:40.13 | [TK]D-Fender | bn43: Welcome to TOASTER-LAND |
| 17:40.15 | angryuser | beherit most of the time it's when asterisk is unable to join the destination, i.e sip peer is no present, but can be many things, you need to read debug as fender said |
| 17:40.30 | [TK]D-Fender | ~users.conf |
| 17:40.30 | jbot | users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 17:40.55 | bn43 | um right - I |
| 17:41.14 | *** join/#asterisk fakhir (i=a7ce8021@gateway/web/ajax/mibbit.com/x-6cc413be58bf5b2c) |
| 17:41.18 | bn43 | I'm getting fairly good at looking and manipulating the actual files now |
| 17:41.42 | bn43 | can I get rid of asterisk-gui without causing nonsense? |
| 17:42.19 | [TK]D-Fender | bn43: Removing GUI's removes nonsense |
| 17:42.57 | *** join/#asterisk otavio_ (n=otavio@debian/developer/otavio) |
| 17:43.11 | *** join/#asterisk hgriffin (n=herman@72.37.252.50) |
| 17:43.15 | hgriffin | hello |
| 17:43.53 | bn43 | once I remove the gui, will it reset my files to defaults or leave them as is? |
| 17:44.14 | hgriffin | I have 2 pseudo channels when I do a 'zap show channels'. Does anyone have a clue why? |
| 17:44.29 | [TK]D-Fender | bn43: No, if you stop using it, the carpet stops being pulled out from under you. |
| 17:44.42 | bn43 | hehe |
| 17:44.56 | bn43 | u got a healthy opinion of the gui |
| 17:44.59 | [TK]D-Fender | bn43: Although since it was built with users.conf in mind I would highly recommend TRASHING just about your entire setup anyways |
| 17:45.27 | angryuser | is there any way to see in console which credentials are used when someone connect's to ami ? (i need the used password) |
| 17:45.28 | [TK]D-Fender | bn43: Yes... I would say "healthy". Don't go picking a PBX you can actually control and make that the first thing you sacrifice |
| 17:45.42 | [TK]D-Fender | angryuser: Nope... uber-patch-worthy |
| 17:45.56 | [TK]D-Fender | hgriffin: MeetMe <- |
| 17:46.20 | [TK]D-Fender | bn43: Never forget to back up your configs anyway |
| 17:46.33 | bn43 | ok I just wanted to ease myself into seeing what asterisk can do and chose the gui to 'see' |
| 17:46.56 | bn43 | but now that I'm familiar I can thrash and do again |
| 17:47.15 | [TK]D-Fender | bn43: Excellent... progress is a process.... |
| 17:47.39 | keebler | Asterisk has nice hold music.. too bad I can't hear anything said from the remote terminal. |
| 17:47.50 | RobH | meh, i forget, what is the sip.conf setting to set aport range for sip rtp traffic and what are the defaults? |
| 17:47.53 | [TK]D-Fender | bn43: In some ways its almost easier coming from FreePBX back to "basic" because they at least break up SIP & VM into sensible segments. |
| 17:48.11 | [TK]D-Fender | bn43: At which point you can concentrate on dialplan with it the real 95% of configuring * |
| 17:48.29 | [TK]D-Fender | robIsn't in sip.conf <- rtp.conf |
| 17:48.49 | keebler | [TK]D-Fender: Do I need to setup port forwarding or listening on the routers/bridges? |
| 17:48.52 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
| 17:49.01 | RobH | ahh, thx! |
| 17:49.04 | bn43 | I followed a howto on installing from source - asterisk. then asterisk-gui from svn - what is the recommended course of action to thrash and start again? |
| 17:49.04 | hgriffin | [TK]D-Fender: Does MeetMe create new pseudo device whenever a conference starts? |
| 17:49.04 | [TK]D-Fender | keebler: ... huh? |
| 17:49.19 | keebler | [TK]D-Fender: Uh. Well. Not sure what I'm doing at this point with asterisk. |
| 17:49.31 | [TK]D-Fender | bn43: just look at the configs that enable HTTP for *, and trash the GUI folder I guess... |
| 17:49.34 | angryuser | keebler: its port forwarding |
| 17:49.45 | [TK]D-Fender | keebler: And I have no idea where you're going with that insanity :0 |
| 17:49.55 | [TK]D-Fender | keebler: What kind of forwarding do you assume I need? :) |
| 17:50.18 | [TK]D-Fender | keebler: Forwarding what? listening for what? |
| 17:50.25 | [TK]D-Fender | keebler: Talking IDS here? |
| 17:50.31 | keebler | I've got two bridges, technically the same subnet. But, one phone on one bridge and the other phone on the other bridge. Do I need to direct the traffic? |
| 17:50.39 | [TK]D-Fender | hgriffin: IIRC yes. |
| 17:50.55 | [TK]D-Fender | keebler: Ah talking about your WifFi bridge? |
| 17:51.01 | keebler | I can make each phone ring from one or the other, but I can't hear any voice. |
| 17:51.04 | keebler | [TK]D-Fender: Yes. |
| 17:51.09 | [TK]D-Fender | keebler: Never did a "same subnet" bridge before. |
| 17:51.13 | bn43 | [TK]D-Fender: but will that restore my configs to defaults? ie as in a virgin installation? |
| 17:51.36 | *** join/#asterisk reneger (n=reneger@dslb-088-078-115-255.pools.arcor-ip.net) |
| 17:51.43 | [TK]D-Fender | bn43: Shouldn't... you're just disabling a daemon and ripping the supoprt bits out |
| 17:51.55 | keebler | Each phone is statically set, as or the bridges. |
| 17:51.58 | [TK]D-Fender | bn43: then again thats why I told you to back up the folder... just in case |
| 17:52.01 | bn43 | I'd like to start from afresh and become a cli junky like u guys |
| 17:52.30 | bn43 | nothing needed to save as I been only testing 2 phones on basic configurations |
| 17:52.33 | *** join/#asterisk hfb (n=hfb@pool-96-229-38-185.lsanca.dsl-w.verizon.net) |
| 17:52.53 | keebler | bn43: Whats stopping you from starting over? |
| 17:52.59 | [TK]D-Fender | bn43: *-GUI almost forces you to because it doesn't do a "complete" job and the bits its missing have you finding you've painted yourself into a corner and the antural expansion reflex has you learning you should have done it yourself in the first place. |
| 17:53.15 | [TK]D-Fender | bn43: Here's some quick "inspiration" : |
| 17:53.17 | [TK]D-Fender | ~jerjerguide |
| 17:53.18 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 17:53.22 | [TK]D-Fender | ^^^^ |
| 17:53.32 | bn43 | keebler: basically I don't know how to start afresh |
| 17:53.39 | keebler | [TK]D-Fender: Thanks for that one BTW, he helped me a ton. |
| 17:53.41 | bn43 | cool - I'll look |
| 17:53.52 | keebler | bn43: Download favorite flavor of *nix. |
| 17:53.58 | keebler | Start from there. |
| 17:54.00 | keebler | :P |
| 17:54.11 | bn43 | I'm running mine on ubuntu |
| 17:54.17 | wonderworld | sup |
| 17:54.28 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:54.32 | bn43 | but I compiled from source - so can't aptitude purge |
| 17:54.43 | bn43 | and reinstall afresh |
| 17:54.54 | keebler | Heh. Delete-reinstall |
| 17:55.04 | manxpower | bn43: To start fresh remove /etc/asterisk /usr/lib/asterisk /var/spool/asterisk |
| 17:55.19 | bn43 | that all?? |
| 17:55.26 | manxpower | anything left over will be overwritten when you reinstall |
| 17:55.34 | bn43 | cool! |
| 17:55.49 | keebler | rm -rf /etc/asterisk |
| 17:56.02 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 17:56.25 | [TK]D-Fender | bn43: Ok, you already know your packaging... You should do just fine... |
| 17:57.29 | *** join/#asterisk FuriousGeorge (n=Brian@ool-4354d18c.dyn.optonline.net) |
| 17:57.39 | keebler | Now that he's got his direction.. anyone have an idea where I should start looking as to why I'm not getting any voice? I can put the line on hold and get music, but can't talk. |
| 17:57.54 | manxpower | keebler: NAT |
| 17:58.15 | keebler | Enable nat on the ATAs or the router? |
| 17:58.21 | n3hxs | ?? nat |
| 17:58.29 | n3hxs | ?? nat |
| 17:58.32 | [TK]D-Fender | manxpower: He's running a same-subnet wireless bridge locally |
| 17:58.34 | keebler | And whats nat going to do? |
| 17:58.39 | keebler | Yeah.. |
| 17:58.43 | [TK]D-Fender | n3hxs: this isn't #freepbx |
| 17:58.44 | keebler | Ah, he didn't know. |
| 17:58.49 | n3hxs | LOL. |
| 17:59.05 | keebler | But yeah, I am on the same subnet. |
| 17:59.15 | keebler | Each phone is on a remote bridge. |
| 17:59.36 | FuriousGeorge | is it safe to say that asterisk 1.6 is "not ready for primetime"? |
| 17:59.38 | keebler | thing triangle with the top being the host, and the two corners being the client bridges/ |
| 17:59.47 | FuriousGeorge | and i should stick with 1.4.2X? |
| 17:59.48 | keebler | think* |
| 17:59.58 | beek | FuriousGeorge: Depends on your definition of "primetime." I'm using it with no problems. |
| 18:00.32 | FuriousGeorge | beek: ive had less than 100% reliability with 1.4.X, so I'm open to change... i just can't let it get worse |
| 18:01.09 | beek | FuriousGeorge: If you're uber-conservative, stay with the 1.4 branch. I've had no show-stoppers for my 1.6 installation and it's been running now for about eight weeks, situated between the PSTN and my legacy PBX |
| 18:01.14 | *** join/#asterisk cp5 (n=samy@cpe-76-171-169-53.socal.res.rr.com) |
| 18:01.16 | cp5 | has anyone ran into asterisk 1.6.0 call file issues? the call file exists, asterisk *sees* it but says "no such file or directory" when opening it |
| 18:01.30 | beek | cp5: Permissions issue? |
| 18:01.34 | FuriousGeorge | i just got a call from a client that in rapid succession an inbound call over zap and an outbound call over iax2 disconnected mid-conversation with a 'pop' |
| 18:01.47 | cp5 | i've confirmed the user that's running asterisk can read it -- i can run: "!cat /.../path/to/call.file" and it works |
| 18:01.48 | cp5 | in asterisk |
| 18:01.58 | cp5 | and can list files in the directory |
| 18:02.03 | [TK]D-Fender | cp5: Show us your complete process |
| 18:02.09 | [TK]D-Fender | pcPASTEBING it all |
| 18:02.12 | cp5 | k |
| 18:02.15 | [TK]D-Fender | ~pb |
| 18:02.16 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 18:02.20 | FuriousGeorge | other servers will sometimes (once every few months) not hangup a zap call... i should probably go back to restarting nightly |
| 18:02.23 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) |
| 18:03.35 | manxpower | FuriousGeorge: using analog? |
| 18:03.41 | cp5 | [TK]D-Fender, just got it working. had to set the gid the same as the uid -- weird |
| 18:03.47 | FuriousGeorge | manxpower: no choice |
| 18:04.02 | [TK]D-Fender | cp5: Alrighty... |
| 18:04.03 | manxpower | FuriousGeorge: in my experience analog cards in Asterisk just randomly lock up. |
| 18:04.05 | cp5 | thanks |
| 18:04.07 | FuriousGeorge | i guess ill wait for 1.6 to be in double digit point release territory |
| 18:04.12 | manxpower | Good thing not everyone experiences this. |
| 18:04.13 | FuriousGeorge | manxpower: mine too |
| 18:04.31 | manxpower | FuriousGeorge: Oh, I've not used analog since early 2003 because of this issue. |
| 18:04.48 | bn43 | ok I've removed those folders - I'm searching for references to asterisk-gui and only find it in /usr/src - do I just delete that folder and I'm rid of any remnants of the gui? |
| 18:05.20 | manxpower | bn43: I can't help with GUIs |
| 18:05.23 | keebler | you could ignore it and don't even worry about installing *-gui. |
| 18:05.35 | bn43 | not going to :-) |
| 18:05.45 | bn43 | just want to remove it from my system |
| 18:05.50 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-46-53.w86-215.abo.wanadoo.fr) |
| 18:06.12 | keebler | If nothing else depends on it, then why should it hurt if you remove? Delete away. :) |
| 18:06.30 | FuriousGeorge | manxpower: i foolishly hold out hope for zaptel 1.4.12.2 |
| 18:07.24 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 18:07.39 | [TK]D-Fender | FuriousGeorge: * dev doesn't work like that now... |
| 18:07.47 | bn43 | right here i go..... |
| 18:08.01 | [TK]D-Fender | FuriousGeorge: 1.6.0.X is a semi-major ver, 1.6.1.X is in the works already with RC's |
| 18:08.15 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 18:10.03 | FuriousGeorge | [TK]D-Fender: thanks for the heads up |
| 18:10.24 | FuriousGeorge | i gotta take off. thanks for the conversation all |
| 18:11.22 | bn43 | um - the make and make installs processed extremely fast this time - should I have untarred afresh in /usr/src? |
| 18:12.22 | bn43 | pardon my ignorance but I have become used to aptitude to install my software - don't come from the ./configure generation! |
| 18:13.14 | jaytee | the ./configure generation? is that an oblique method of calling us all dinosaurs? |
| 18:13.28 | bn43 | nope - paying respects...... |
| 18:13.30 | jaytee | Raptors!!! To the attack!!!! |
| 18:13.34 | bn43 | :-) |
| 18:13.56 | beek | bn43: Don't smile -- jaytee can call the raptors! |
| 18:14.09 | bn43 | apologies! |
| 18:14.21 | manxpower | Is there a GOSUB_EXTEN like MACRO_EXTEN? |
| 18:14.23 | bn43 | kind sirs...have mercy.... |
| 18:14.24 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
| 18:14.30 | jaytee | I'm not only older than dirt, the day after dirt was invented I invented the shovel, stupidly I made it open source :-) |
| 18:14.59 | manxpower | bn43: your problem is that all the GUI users are, strangely enough on #asterisk-gui |
| 18:15.11 | beek | manxpower: I haven't found one. I end up using ${EXTEN} in the call to the gosub. |
| 18:15.11 | jaytee | manxpower, don't recall seeing that in the channel variables text file |
| 18:15.33 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 18:15.33 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 18:15.44 | jaytee | what about using _ to make it inherit like in macros? |
| 18:15.55 | bn43 | yes but I am trying to correct the error of my ways - somehow I think going to asterisk-gui will not endear me to them |
| 18:16.02 | bn43 | so am I ok? |
| 18:16.15 | [TK]D-Fender | ${MACRO_EXTEN} = lazy. real men pass it like the ARG it deserves to be |
| 18:16.16 | manxpower | bn43: you're already ruined your system. |
| 18:16.29 | bn43 | that bad? |
| 18:16.32 | manxpower | [TK]D-Fender: and yet Gosub does not support ARGs in 1.4 |
| 18:16.34 | [TK]D-Fender | bn43: You're doing fine, just keep trudging through |
| 18:16.46 | [TK]D-Fender | manxpower: So you should be using MACRO. |
| 18:17.08 | [TK]D-Fender | manxpower: Get off that fence or I'll have Vlad sharpen the ends for you :p |
| 18:17.09 | beek | manxpower: I like passing it in as an argument to make it clear what I'm doing. |
| 18:17.26 | manxpower | [TK]D-Fender: macro is going away |
| 18:17.33 | manxpower | beek: I think your way is what I'll use. |
| 18:17.45 | sigmounte | [TK]D-Fender, thanks for your help , indeed disa is doing something stupid and don't allow Monitor to work |
| 18:17.59 | jaytee | well, if macro is going away then whatever replaces it better be as good and frankly, Gosub ain't it. |
| 18:18.33 | [TK]D-Fender | sigmounte: How NOT to use DISA : go record a LOT of dialtone and BACKGROUND it in an IVR. Same effect, less BS |
| 18:18.58 | [TK]D-Fender | jaytee: GOSUB has been hybridized in 1.6 |
| 18:19.24 | jaytee | hybridized? you mean the gene spliced it with some corn? |
| 18:19.28 | manxpower | jaytee: as I understand it 1.6's Gosub is pretty good. |
| 18:19.32 | jaytee | s/the/they |
| 18:19.46 | jaytee | I don't have a 1.6 box handy to do a core show |
| 18:19.47 | beek | jaytee * manxpower : I'm quite happy with it. |
| 18:20.27 | manxpower | Virtually all of my stuff is 1) Set variables 2) run macro/gosub so I don't really need Gosub args (except for EXTEN) |
| 18:20.48 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 18:21.20 | manxpower | [TK]D-Fender: I'm finally rewriting some of my macros |
| 18:22.11 | *** part/#asterisk ftp3 (n=none@pool-71-117-187-57.ptldor.dsl-w.verizon.net) |
| 18:22.19 | [TK]D-Fender | manxpower: Yeah that last mostrosity of yours.... *shudder* |
| 18:24.22 | manxpower | [TK]D-Fender: the new version should be just ugly rather then hideous |
| 18:24.33 | [TK]D-Fender | manxpower: IMPRESSIVE! |
| 18:26.05 | *** join/#asterisk km- (n=pgrace@2001:470:8a93:2:0:0:0:2) |
| 18:26.28 | km- | can iaxmodem be used with minicom or seyon to use for remote access? |
| 18:26.42 | km- | all I see mentioned on iaxmodem forums are fax, fax, fax |
| 18:27.30 | km- | we have a purely voip setup going to a PRI, hoping that we might use something like iaxmodem to allow us to do remote console for a different site. |
| 18:28.27 | rue_mohr | so to be clear, the oslec drivers require you to collect source from kernel source and modify a few files, and it dosn't work. what is oslec used for if not for asterisk? |
| 18:28.33 | *** join/#asterisk hardwire (n=hardwire@srv001.gandi.brutetech.com) |
| 18:28.35 | hardwire | y0 |
| 18:28.54 | cp5 | where are there docs on state_interface in 1.6? anything i've seen says to search for "asterisk queue state" on the asterisk-users list, and when i search google for the archive and that text, all i can find is an email saying to search for that :\ |
| 18:29.42 | rue_mohr | and the dahdi drivers have provisions for oslec that are useless? |
| 18:30.41 | manxpower | [TK]D-Fender: using something that tries to be a programming language will make things much easier. That and getting rid of all the 1.0isms and 1.2isms. |
| 18:31.48 | [TK]D-Fender | manxpower: taht too.. just seriously... AEL... don't |
| 18:31.48 | jaytee | i hate isms |
| 18:32.02 | rue_mohr | infinite state machines!? |
| 18:32.35 | [TK]D-Fender | rue_mohr: In order to form a more perfect union... |
| 18:37.57 | [TK]D-Fender | jaytee: I avoid cliches like the plague |
| 18:38.20 | *** join/#asterisk fakhir (i=a7ce8021@gateway/web/ajax/mibbit.com/x-e986be437e56141a) |
| 18:39.40 | *** join/#asterisk CapriCoRN^80 (n=int@207.176.6.160) |
| 18:41.14 | *** join/#asterisk eppigy (n=Dave@plasticlobster.com) |
| 18:41.20 | eppigy | SMOKE PURP BY THE POUND |
| 18:41.45 | bn43 | i'm following jeremy's howto on a basic config - I find that the files he refers to already have contents and some lines are uncommented/active - should I delete those files and only have his contents in the files? |
| 18:42.37 | [TK]D-Fender | bn43: Read the sample to understand the structures that were created and why |
| 18:42.44 | [TK]D-Fender | bn43: then build your own. |
| 18:43.08 | *** join/#asterisk icebrew54 (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 18:45.06 | eppigy | hello [TK]D-Fender |
| 18:45.13 | eppigy | i am glad to see you are well |
| 18:45.16 | [TK]D-Fender | eppigy: You are dave |
| 18:45.22 | eppigy | yes |
| 18:45.32 | eppigy | this statement proves to be factual |
| 18:46.06 | icebrew54 | using asterisk-gui's trunkdial-failover-0.3 and am getting a "Congestion" error message on attempting to dial out via 2nd channel....primary trunk is PSTN/Zaptel, secondary is SIP |
| 18:46.19 | icebrew54 | can anyone suggest troubleshooting methodology on this? |
| 18:48.13 | [TK]D-Fender | icebrew54: Go look at actual CLI output, verbose 10 w/ whatever channel debug is appropriate to what you're calling. |
| 18:49.43 | icebrew54 | [TK]D-Fender: it's attempting to dial out via Zaptel/g1 twice |
| 18:49.49 | icebrew54 | [TK]D-Fender, [Jan 27 10:41:58] WARNING[28557]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) |
| 18:49.53 | *** part/#asterisk km- (n=pgrace@2001:470:8a93:2:0:0:0:2) |
| 18:50.02 | icebrew54 | at this point it should hit the SIP trunk |
| 18:50.10 | [TK]D-Fender | icebrew54: In some ISDN implementations that can actually mean the CALLEE is "busy" |
| 18:50.33 | icebrew54 | using digium PSTN 410p (analog) card |
| 18:50.41 | [TK]D-Fender | icebrew54: And if you think it should do something more, then go look at your dialplan. |
| 18:50.41 | icebrew54 | and that line (g1) was in use during that time |
| 18:50.51 | icebrew54 | ok...can you suggest reading for "failover" trunk dialing? |
| 18:51.31 | [TK]D-Fender | icebrew54: No such thing. Its all just dilaplan. Go look at what you're DOING |
| 18:51.38 | [TK]D-Fender | dialplan* |
| 18:52.28 | icebrew54 | ok, understood |
| 18:56.21 | *** join/#asterisk chi6IT41 (n=chigital@tmo-105-134.customers.d1-online.com) |
| 19:00.05 | bmoraca | yay for GUIs, lol |
| 19:01.26 | eppigy | no |
| 19:01.41 | eppigy | thats like saying yay for downs syndrome |
| 19:01.53 | [TK]D-Fender | \o/ |
| 19:02.12 | eppigy | YAY FOR AUTISM |
| 19:02.38 | bmoraca | i was being facetious...but, yeah |
| 19:02.57 | eppigy | calm down killer |
| 19:04.21 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 19:05.24 | [TK]D-Fender | Know whats really funny? Watching clueless GUI users try to help other equally clueless GUI users :) |
| 19:05.40 | [TK]D-Fender | Run in circles loking at crap that doesn't matter... |
| 19:06.02 | eppigy | agreed |
| 19:06.09 | hardwire | does ztd-eth imply ztdummy? or will you still need ztdummy if you don't have a valid timing source. |
| 19:06.19 | bmoraca | i would agree...if you don't understand what's going on behind the GUI, then you shouldn't be using the GUI |
| 19:06.59 | [TK]D-Fender | bmoraca: No... if you don't understand much of anything you should refrain from "helping" people a bit :p |
| 19:07.52 | bmoraca | well, that too |
| 19:09.00 | *** join/#asterisk dmz (n=dmz@146.sub-70-212-145.myvzw.com) |
| 19:09.41 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 19:09.41 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:10.03 | eppigy | be cool russellb's here |
| 19:10.15 | *** join/#asterisk xpat (n=xp@206-248-174-2.dsl.teksavvy.com) |
| 19:10.38 | russellb | heh |
| 19:10.59 | [TK]D-Fender | dumps eppigy into a vat of liquid nitrogen |
| 19:11.10 | russellb | ~thwack [TK]D-Fender |
| 19:11.10 | jbot | ACTION hits [TK]D-Fender on the thumb with a 5ESS Switch |
| 19:11.17 | Qwell | pwnt |
| 19:11.21 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 19:11.38 | xpat | I'm looking for a current master list of sounds for Asterisk (with transcriptions hopefully). Does such a list exist? |
| 19:11.55 | Qwell | xpat: yes. in the sounds tarball |
| 19:11.57 | [TK]D-Fender | russellb: Already been sliced open by a sword, do you think a bruise'll faze me? hah! |
| 19:12.00 | Qwell | go figure.. |
| 19:12.09 | xpat | hmm, think I would have checked there first... |
| 19:12.11 | xpat | OK thanks!! |
| 19:12.12 | eppigy | TRABJAO |
| 19:12.23 | [TK]D-Fender | eppigy: TYOP |
| 19:12.46 | eppigy | QUE? |
| 19:13.13 | [TK]D-Fender | </sarcasm> |
| 19:13.57 | jaytee | TRABJAO? don't you mean TRABAJO? |
| 19:14.24 | eppigy | jaytee: SI GRACIAS MUCHACHO |
| 19:15.46 | rob0 | JOATBAR (jack of all trades beyond all recognition) |
| 19:15.49 | [TK]D-Fender | Today's deep though : Do dyslexics with Tourettes shoot people at random? |
| 19:16.50 | *** join/#asterisk mahlon (i=mahlon@martini.nu) |
| 19:18.08 | sigmounte | Anyone know where i can find soxmix ? (i've downloaded the sox source and not soxmix present in it ? ) |
| 19:18.14 | jaytee | if i set pri debug file to a file do I need to go and set pri debug span (some span3) or does just setting pri debug file /someplace/somefile.txt initiate the pri debug? |
| 19:18.22 | Qwell | sigmounte: it's part of the main sox binary, using a switch |
| 19:18.33 | sigmounte | a switch ? |
| 19:18.44 | Qwell | man sox |
| 19:19.24 | eppigy | jaytee: you must set the span to debug |
| 19:19.29 | Corydon76-dig | Qwell: or a hardlink |
| 19:19.44 | jaytee | man woman |
| 19:19.49 | sigmounte | hmm, i use a script who use somix directly , i'll have to add the switch to it , i hope it's compatible |
| 19:19.53 | jaytee | No manual entry for woman |
| 19:19.58 | jaytee | rats |
| 19:20.24 | *** join/#asterisk plc5_250 (n=ron@c-68-40-223-224.hsd1.mi.comcast.net) |
| 19:20.58 | plc5_250 | Hi everyone - I need some help with a non-working MWI between * 1.4 and a SNOM 220 |
| 19:23.13 | plc5_250 | is anyone from Digium support on here? |
| 19:26.36 | *** join/#asterisk chi6IT41 (n=chigital@tmo-105-134.customers.d1-online.com) |
| 19:27.00 | Qwell | plc5_250: What is your question? |
| 19:27.26 | plc5_250 | I can't get MWI to work between a SNOM220 and *1.4.17 |
| 19:27.36 | Qwell | What does that have to do with Digium support? |
| 19:27.49 | plc5_250 | digium == *, correct? |
| 19:28.10 | Qwell | So, you want free Digium support for an Open Source program? |
| 19:28.30 | plc5_250 | digium directs people to this resource for support |
| 19:28.52 | Qwell | I think you may have failed to see the reasoning. |
| 19:29.06 | Qwell | It doesn't say "Go here and Digium will give you free support." |
| 19:29.10 | *** join/#asterisk moy (n=chatzill@bas1-unionville55-1177733953.dsl.bell.ca) |
| 19:29.13 | keebler | If nothing else depends on it, then why should it hurt if you remove? Delete away. :) |
| 19:29.18 | keebler | Oops |
| 19:29.20 | bn43 | mmm I followed the tutorial but I think I may have stuffed up somewhere - the phone says error 404 on the logs and the console says "no matching peer found" |
| 19:29.30 | keebler | What is the likelihood that flashing two WRT54G v6's would produce two routers with the exact same MAC Addresses? |
| 19:29.33 | bmoraca | keebler: lol |
| 19:29.45 | Qwell | keebler: How many in 48 bits? |
| 19:29.49 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 19:29.54 | bn43 | how do I list the peers again? |
| 19:30.07 | bmoraca | bn43: sip show peers |
| 19:30.42 | bn43 | yeah I should have just typed help! |
| 19:30.59 | bn43 | says not found - mmm - weird |
| 19:31.13 | keebler | Qwell: How many what? |
| 19:31.21 | keebler | Qwell: All three of them are the same. |
| 19:31.40 | keebler | 00:40:10:10:00:01/02/03 |
| 19:31.42 | manxpower | keebler: Using the sveasoft stuff? |
| 19:31.46 | keebler | DDWRT |
| 19:31.58 | manxpower | Dunno about that, but sveasoft does something with your MAC |
| 19:32.17 | keebler | I've got a WRT54g V2.2 and its got its original MAC iirc. |
| 19:32.19 | plc5_250 | keebler - MAC addresses are usually stored in a PROM as a part of the ethernet circuitry |
| 19:32.22 | manxpower | are the MACs on the bottom of the router the same? |
| 19:32.44 | bmoraca | keebler: MAC addresses have a specific structure, they're not random. The first 24 bits are the manufacturer. the next ones are dependent on the device and serialization. |
| 19:33.31 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 19:33.54 | manxpower | Many devices (especially consumer routers) let you override the system MAC with one you specify |
| 19:34.10 | keebler | manxpower: no |
| 19:34.20 | eppigy | FINISH HIM |
| 19:34.33 | plc5_250 | that's usually done via software - letting the software driver handle the MAC decisions. |
| 19:36.51 | [TK]D-Fender | Same thing |
| 19:37.04 | [TK]D-Fender | perception = reality |
| 19:37.20 | [TK]D-Fender | goes to install his Heisenberg Compensators |
| 19:37.21 | bmoraca | keebler: it appears that DDWRT uses the 00:40:10 mac scope quite a bit. the 01, 02, 03 is just serialization. |
| 19:37.25 | manxpower | [TK]D-Fender: You have apparently never used LSD |
| 19:37.45 | *** join/#asterisk Micho123 (n=mcho123@63.216.126.129) |
| 19:37.45 | [TK]D-Fender | manxpower: You apparently haven't done it with Incredible Cosmic Powers :p |
| 19:37.56 | keebler | bmoraca: crap, well that sucks. Because I can't bridge two identical routers together. |
| 19:37.58 | manxpower | [TK]D-Fender: true 8- |
| 19:38.07 | manxpower | keebler: so CHANGE the mac |
| 19:38.23 | keebler | manxpower: Yeah, I'm just pissed cause I shouldn't HAVE to .ahah |
| 19:38.25 | manxpower | heck, I'd change them back to whatever is on the sticker on the box |
| 19:38.25 | bmoraca | keebler: you can change the MAC. and if they're 01, 02, and 03, then they're not identical. |
| 19:38.30 | manxpower | keebler: I agree with that. |
| 19:38.39 | manxpower | I like keeping the MAC correct. Helps down the road |
| 19:38.41 | plc5_250 | be careful when setting 2 interfaces to the same MAC, some switches really don't like seeing that happen |
| 19:38.59 | bmoraca | keebler: unless you're saying that each of the three interfaces (wan, lan, wlan) takes 01, 02, and 03 respectively |
| 19:39.02 | keebler | bmoraca: The device has three. |
| 19:39.06 | keebler | yeah |
| 19:39.07 | bmoraca | yeah |
| 19:39.11 | Micho123 | Hi all, i was trying to enable t.38 om my asteriks 1.4.22.1...I installed asterisk-trunk and compile ie...when trying to restart asterisk I got the following error...http://pastebin.com/d3eac2ebe |
| 19:39.20 | *** join/#asterisk snaud (i=lp@hades.ds1.agh.edu.pl) |
| 19:39.20 | Micho123 | can someone help please? |
| 19:39.24 | *** join/#asterisk obnauticus (n=lol@about/windows/regular/obnauticus) |
| 19:39.43 | manxpower | Micho123: you either installed 1.4.22.1 or you installed trunk. Which is it? |
| 19:40.03 | Micho123 | manxpower, I had asterisk installed...Just instaled asterik trunk |
| 19:40.22 | Micho123 | manxpower, asterisk was running smoothly |
| 19:41.28 | bn43 | I'm really puzzled here - extensions.conf shows the stations and details but I still get error no peer |
| 19:41.43 | *** join/#asterisk unpaidbill (n=bill@alteredbeastiality.org) |
| 19:42.16 | hardwire | is ztd-tdm and zaptel support multi-span dynamic tdmoe by default? |
| 19:42.28 | hardwire | that's a big yes! |
| 19:43.00 | *** join/#asterisk stevetotaro (n=Steve@pool-71-254-231-87.hrbgpa.east.verizon.net) |
| 19:43.12 | kaldemar | bn43: peers are not defined in extensions.conf. extensions.conf is the dialplan. |
| 19:43.27 | bn43 | doh |
| 19:44.02 | [TK]D-Fender | bn43: sip.conf <- |
| 19:44.11 | [TK]D-Fender | bn43: (hoping you've now ditched users.conf) |
| 19:45.42 | bn43 | um - I have put in the details at the end of the sip.conf file |
| 19:45.43 | *** join/#asterisk rwaite (n=fieldyca@rrcs-74-218-125-86.central.biz.rr.com) |
| 19:46.41 | bn43 | the only thing I changed was host=static as the phones have a static address |
| 19:47.25 | [TK]D-Fender | bn43: no |
| 19:47.39 | [TK]D-Fender | bn43: Let your phones register like normal. that isn't a valid value anyways |
| 19:47.50 | [TK]D-Fender | bn43: it would have been "host=theiphere" |
| 19:47.56 | [TK]D-Fender | bn43: But seriously... don't |
| 19:48.05 | [TK]D-Fender | bn43: Unless you have DHCP spanning issues of course. |
| 19:48.13 | [TK]D-Fender | bn43: at which point follow the above format |
| 19:48.53 | bn43 | I'm not following - I'm not running a dhcp server - is that recommended? |
| 19:49.19 | bmoraca | bn43: doesn't matter. |
| 19:49.40 | [TK]D-Fender | bn43: thats fine... unsual and tends to show people set duplicate IP's and other sillyiness but you may require it so. |
| 19:49.55 | [TK]D-Fender | bn43: "host=1.2.3.4" <- |
| 19:50.57 | bn43 | aha! |
| 19:51.17 | bn43 | ita now working! |
| 19:51.58 | bn43 | awesome |
| 19:52.29 | bn43 | it actually is easy to set up a basic system |
| 19:52.42 | *** join/#asterisk andrewct (i=andrewct@vpnclient2.ntplx.net) |
| 19:53.19 | *** join/#asterisk Gido-E (n=gido@lounge.datux.nl) |
| 19:53.22 | [TK]D-Fender | bn43: Quite |
| 19:53.25 | andrewct | Anyone from digium here that can help with chan_sip ? |
| 19:53.44 | [TK]D-Fender | andrewct: Ask a specific question, you might get a specific answer... |
| 19:54.36 | Kobaz | schweet |
| 19:54.40 | Kobaz | voicepulse gives me callerid name now |
| 19:54.46 | rwaite | asterisk <-> NAT router <-> internet <-> NAT router <-> iax softphone (zoiper) |
| 19:54.57 | andrewct | With SIP authentication, if I have a conflicting user as a phone number, then an incoming call with the same number from another gateway fails because it does not use that username for authentication. |
| 19:55.07 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 19:55.09 | rwaite | why would the ip show as the nat router's ip when registering? |
| 19:55.59 | Kobaz | rwaite: because of nat of course |
| 19:56.20 | rwaite | will this cause a problem? |
| 19:56.31 | Kobaz | if your configuration is borked, it would |
| 19:56.34 | andrewct | If I rewrite chan_sip to do peer authentication first then there seem to be some issues with username authentication, but it solves the ip peer gateway issue. |
| 19:56.37 | rwaite | hmm. |
| 19:57.08 | Micho123 | manxpower, still around? |
| 19:57.13 | [TK]D-Fender | andrewct: 6 of 1 , half-dozen of the other... |
| 19:57.20 | *** join/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 19:57.34 | andrewct | Once a user is authenticated in chan_sip, does asterisk add it's IP to the peer list? Thus causing the problem with future authentications. |
| 19:58.33 | [TK]D-Fender | andrewct: thats normally "insecure=port,invite" territory |
| 19:58.42 | mikealeonetti | does users.conf work with every asterisk version? |
| 19:59.01 | andrewct | True, but because of the matching user it fails authentication and does not match the insecure peer. |
| 19:59.33 | Kobaz | you may need fromuser= in the sip peer |
| 20:00.09 | andrewct | But the from user is different because it's a PRI gateway (cisco AS5300) and the from is the PSTN caller ID. |
| 20:00.14 | bmoraca | mikealeonetti: only if it's included in sip.conf |
| 20:00.31 | mikealeonetti | bmoraca: so is pretty much the same thing as sip.conf? |
| 20:00.36 | mikealeonetti | in that it defines more sip channels? |
| 20:00.38 | andrewct | This is the old problem of "don't name your users 10 digit phone numbers" |
| 20:00.44 | mikealeonetti | or users rather |
| 20:00.51 | bmoraca | mikealeonetti: yes. |
| 20:00.55 | mikealeonetti | okay I see |
| 20:00.55 | mikealeonetti | thanks |
| 20:02.02 | [TK]D-Fender | andrewct: SIP = Fun (if you LIKE pain that is) |
| 20:02.14 | [TK]D-Fender | ~users.conf |
| 20:02.15 | jbot | users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 20:02.16 | [TK]D-Fender | ^^^^^^^^^^ |
| 20:02.29 | [TK]D-Fender | mikealeonetti: Only warning : Don't |
| 20:02.36 | bmoraca | lol |
| 20:02.39 | mikealeonetti | lol |
| 20:02.50 | mikealeonetti | well, I'm trying to set up this ADA demo |
| 20:03.06 | mikealeonetti | and the config told me to edit my users.conf so I was like "my what...?" |
| 20:03.22 | [TK]D-Fender | ADA? |
| 20:03.38 | [TK]D-Fender | Oh God... *THE* Toaster |
| 20:03.42 | mikealeonetti | Asterisk Desktop Assistant (beta) |
| 20:03.47 | andrewct | Asterisk SIP = Problems with authentication |
| 20:03.57 | mikealeonetti | why are we talking about Cylons? |
| 20:03.59 | bmoraca | wow |
| 20:04.19 | mikealeonetti | (that was funny to me...) |
| 20:05.11 | *** join/#asterisk op3r (n=op3r@ded-139-109.eglobalreach.net) |
| 20:06.03 | mikealeonetti | [TK]D-Fender: what do you mean by "THE Toaster?"' |
| 20:06.16 | bmoraca | mikealeonetti: It looks like that's meant to be used with AsteriskNOW which uses asterisk-gui which uses users.conf. |
| 20:06.27 | hardwire | anybody set up RBS w/ zaptel? |
| 20:06.57 | bmoraca | AsteriskNOW is a toaster. a pretty crappy toaster by comparison to some of the other toasters. there are toasters that can make egg mcmuffins in one pass...complete with a poached egg. |
| 20:07.05 | mikealeonetti | lol |
| 20:07.26 | mikealeonetti | bmoraca: but surely it can be used with all versions of astiersk, no? |
| 20:07.39 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-211-250-236.dsl.pltn13.sbcglobal.net) |
| 20:07.51 | op3r | should I worry about this error? [Jan 27 12:40:20] WARNING[5734]: translate.c:163 framein: no samples for ulawtolin I get this alot on the cli then my asterisk crash |
| 20:07.54 | bmoraca | i would imagine. i've never looked at it before, except to see a press release that mentioned both it and asterisknow in the same paragraph |
| 20:08.08 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 20:08.30 | manxpower | op3r: the message is caused BY the crash, the message does not CAUSE the crash |
| 20:08.40 | *** join/#asterisk n3hxs (n=HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 20:08.42 | manxpower | hardwire: what RBS stuff do you need? |
| 20:09.05 | op3r | manxpower: so any way to find out the cause? |
| 20:09.12 | *** join/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 20:09.13 | *** join/#asterisk `bensh (n=Ben@119.93.45.181) |
| 20:09.18 | manxpower | op3r: not from that message. |
| 20:09.38 | mcargile | what is the option in iax.conf to change the time between registers? I cannot find it in the config example |
| 20:09.39 | manxpower | you can compile asterisk with debug, do a backtrace, etc. all should be listed in backtrace.txt in the Asterisk source. |
| 20:10.04 | Micho123 | Hi all, iguess that asterisk-trunk is not compatible with asterisk 1.4...Is that correect? |
| 20:10.32 | *** join/#asterisk citywok (n=chatzill@corpnet.csgopenline.com) |
| 20:11.40 | Micho123 | If I need to enable T.38 on asterisk then should I install asterisk-trunk or not? |
| 20:12.05 | hardwire | manxpower: experience, mostly. |
| 20:12.06 | hardwire | :P |
| 20:12.58 | *** join/#asterisk ocnarf (n=chatzill@125.252.90.5) |
| 20:13.08 | *** join/#asterisk daniev (n=dnv@190.144.60.154) |
| 20:13.21 | ocnarf | What does "no samples for ulawtolin" means? |
| 20:13.30 | ocnarf | Im getting this msg, all the time |
| 20:13.31 | hardwire | failed to get samples |
| 20:13.46 | ocnarf | samples? |
| 20:14.03 | manxpower | hardwire: well the only time you care about RBS is on PRI (won't work with PRI) or when using the Zaptel/DAHDI DACS festures. |
| 20:14.30 | ocnarf | This error to be exact: WARNING[25732]: translate.c:163 framein: no samples for ulawtolin |
| 20:14.36 | manxpower | Micho123: there are several levels of T.38 in Asterisk. Which one do you want? |
| 20:14.49 | Micho123 | manxpower, pass thru is OK |
| 20:15.05 | manxpower | ocnarf: That means "Asterisk crashed, but I didn't realize it until I tried to get audio and there was none" |
| 20:15.21 | manxpower | Micho123: then 1.4 should work just fine. |
| 20:15.32 | manxpower | You can use 1.4 or 1.6 no reason to use trunk\ |
| 20:15.41 | Micho123 | manxpower, I see |
| 20:16.11 | manxpower | BEFORE 1.4 was released then you could only get that feature from -trunk |
| 20:16.30 | ocnarf | manxpower: what do you mean by that? |
| 20:16.57 | manxpower | ocnarf: What I mean is what I said earlier. The error message means nothing because asterisk crashed. |
| 20:17.11 | hardwire | manxpower: I need to trunk to some RBS voice T1's |
| 20:17.27 | hardwire | and I see the DACS stuff in zapata.conf |
| 20:17.27 | manxpower | ocnarf: It is like you saying you are seeing a bright light when you are dieing. Does the bright light cause you to die? |
| 20:17.35 | hardwire | but I know that's not the right thing. |
| 20:18.04 | ocnarf | manxpower: ok, got your point. |
| 20:18.06 | manxpower | hardwire: do you want asterisk to process the calls or not? |
| 20:18.11 | hardwire | yes |
| 20:18.20 | hardwire | I'll be doing monitoring/recording with whisper |
| 20:18.25 | manxpower | hardwire: then do it EXACTLY as you would with non-RBS channels |
| 20:18.39 | hardwire | as fxs? |
| 20:18.52 | manxpower | hardwire: however you would do it excluding RBS |
| 20:19.09 | manxpower | you only care about the RBS stuff if you are doing actual DACS |
| 20:19.14 | hardwire | manxpower: yeh.. working as an active pass-thru |
| 20:19.21 | manxpower | otherwise Asterisk can't tell the difference. |
| 20:19.44 | hardwire | I need to make sure I'm RBS in from the CPE and RBS out to the NET |
| 20:20.30 | manxpower | hardwire: robbed bit signaling == CAS == voice T-1 |
| 20:20.49 | hardwire | and the call information is stored in the robbed bits aye |
| 20:20.50 | manxpower | hardwire: have you actually tried it? |
| 20:20.53 | hardwire | like e&m |
| 20:20.58 | manxpower | no, the signaling is in the robbed bits |
| 20:21.02 | hardwire | manxpower: I don't have the equipment here, unfortunately. |
| 20:21.15 | hardwire | I'm more used to E&M and PRI |
| 20:21.26 | manxpower | hardwire: my statements stand unless there is something you don't know. |
| 20:21.36 | manxpower | like it wants a 56K channel |
| 20:21.44 | hardwire | ah. |
| 20:21.49 | hardwire | so what signalling is it? |
| 20:21.54 | hardwire | fxo/fxs? |
| 20:22.03 | hardwire | that's as close to pass-thru as possible right? |
| 20:22.14 | manxpower | I can't know that unless we know what the far end is expecting |
| 20:22.31 | manxpower | hardwire: the only real "passthru" on a T-1 is DACS. |
| 20:22.39 | hardwire | all they are telling me is RBS with jb7/d4 |
| 20:22.58 | hardwire | manxpower: I don't mean passthru in that design. |
| 20:23.04 | hardwire | I meant more for the audio of the call. |
| 20:23.08 | hardwire | since the bits are in the audio |
| 20:23.32 | Micho123 | manxpower, One question...Can i specify a codec for T.38 or the codec is just defined for the extension? |
| 20:23.50 | beherit | is it possible to restrict * to allow only one user at a time can use the extension? right now multiple user can use same ext number. |
| 20:23.53 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 20:24.31 | beherit | let say me and my friend can use the extesion 6060 at the same time just use a seperate machine |
| 20:24.38 | hardwire | trying to get more info from client |
| 20:25.45 | [TK]D-Fender | beherit : use the "GROUP_COUNT" function for this |
| 20:26.26 | *** join/#asterisk chi6IT41 (n=chigital@tmo-103-81.customers.d1-online.com) |
| 20:28.33 | bn43 | I'm having a weird problem now - one of the phones shows NR and the console says wrong password on registration attempt - double checked passwords and its fine - this is straight after testing voicemail |
| 20:29.03 | rue_mohr | only one? |
| 20:29.09 | bn43 | yeah |
| 20:29.15 | rue_mohr | same model phones? |
| 20:29.20 | bn43 | yup |
| 20:29.45 | eppigy | KILL YOUR FAMILY |
| 20:30.19 | bn43 | btw I'm a noob and followed jeremy maknamara's tutorial to set up my syste, |
| 20:30.22 | bn43 | system |
| 20:30.26 | eppigy | wrong window |
| 20:30.37 | rue_mohr | just a sec eating lunch |
| 20:30.57 | Juggie | anyone know of a working free sms gateway? |
| 20:31.05 | hardwire | manxpower: I'm better now |
| 20:31.38 | beherit | [TK]-Fender: is that in sip.conf or extension.conf? |
| 20:32.17 | bn43 | ok - false alarm - phone lost its settings - went to the web interface to double check |
| 20:32.53 | [TK]D-Fender | beherit :extensions.conf |
| 20:33.00 | manxpower | beherit-: An extension is just a line in extensions.conf. everything else is device, callerid, did, etc |
| 20:33.28 | beherit | ok thanks |
| 20:39.54 | andrewct | Is there a programmer from digium here? |
| 20:40.21 | *** part/#asterisk FuriousGeorge (n=Brian@ool-4354d18c.dyn.optonline.net) |
| 20:41.39 | manxpower | does AEL2 care what case the applications are in? |
| 20:42.08 | [TK]D-Fender | manxpower: Shouldn't... extensions.conf doesn't |
| 20:42.32 | rue_mohr | WARNING: "oslec_create" [/usr/src/dahdi-linux-2.1.0.3/drivers/dahdi/dahdi_echocan_oslec.ko] undefined! <- thats provided by the kernel module, right? |
| 20:46.23 | rue_mohr | why do i get the feeling i'm standing in the middle of a desert asking which way to the library, and there's nobody around for miles? |
| 20:46.34 | hardwire | manxpower: have you use dacs as passthru before? it seems pretty awesome. |
| 20:46.57 | manxpower | hardwire: I did quite a bit of DACS |
| 20:47.04 | *** join/#asterisk Micc (n=Michael@c-76-121-255-52.hsd1.wa.comcast.net) |
| 20:47.17 | hardwire | manxpower: did? |
| 20:47.20 | manxpower | We passed some data channels thru asterisk |
| 20:47.28 | *** join/#asterisk bijit (n=benji@190.241.157.5) |
| 20:47.40 | hardwire | I'll have to play with it at some point |
| 20:47.43 | manxpower | hardwire: I started at working a real job a few weeks ago. no more of my old customers |
| 20:47.44 | bijit | when I have this error "chan_sip.c: username mismatch, have <306>, digest has <>" |
| 20:47.54 | hardwire | I'm just a consultant atm and haven't made enough moolah to get his hands on t1 cards yet. :) |
| 20:47.54 | *** join/#asterisk ghenry (n=ghenry@92.41.193.67.sub.mbb.three.co.uk) |
| 20:47.56 | manxpower | also, we switched to PRI, got the data off the T and never needed it again |
| 20:47.58 | bijit | where can i start looking for the error? |
| 20:48.11 | *** join/#asterisk KU0N (n=kuon@119-193.104-92.cust.bluewin.ch) |
| 20:48.15 | hardwire | manxpower: yeh.. I hope this company moves to PRI |
| 20:48.27 | Micc | Is there any drawback to setting the registry timeout to like 2 minutes instead of 3600 seconds? |
| 20:48.47 | Micc | If a phone lags out I want it to reconnect as soon as possible. |
| 20:49.07 | Micc | I like the "dial without reg" option on the linksyss' |
| 20:49.46 | manxpower | Micc: not really. once you go 60 seconds you should punch thru any NAT translations |
| 20:50.30 | [TK]D-Fender | manxpower: 60sec pretty much kills off the need for qualify, thats for sure :) |
| 20:52.17 | tzafrir_laptop | rue_mohr, "oslec_create" should be provided by the module oslec |
| 20:53.39 | *** join/#asterisk sah-work (n=Bawbatos@adsl-76-211-250-236.dsl.pltn13.sbcglobal.net) |
| 20:55.06 | *** join/#asterisk lilkid (n=chatzill@87-194-38-230.bethere.co.uk) |
| 20:55.17 | *** join/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 20:56.11 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 20:57.41 | lilkid | How can I go about managing minutes used/billing etc. ? Easiest way just to install a billing package? -I don't particularly have time to write a whole billing system |
| 21:04.48 | [TK]D-Fender | lilkid: Lookup a2billing on the WIKI |
| 21:04.53 | [TK]D-Fender | ~wikis |
| 21:05.08 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 21:05.30 | Micho123 | Hi all, is there a way to auto select codec when trying to send a FAX? |
| 21:06.16 | *** part/#asterisk `bensh (n=Ben@119.93.45.181) |
| 21:07.37 | lilkid | thanks d-fender |
| 21:08.08 | manxpower | Micho123: faxes only work with ulaw, alaw, and T.38 (in theory) |
| 21:09.42 | bn43 | Hi - I'm encountering a problem with one of my phones - it just loses the password and I have to go in the web interface and put it in again |
| 21:10.02 | [TK]D-Fender | bn43: Your phone sucks. Kick it in the nads. |
| 21:10.13 | bn43 | lol! |
| 21:10.24 | Micho123 | manxpower, thanks |
| 21:10.30 | bn43 | but it wasn't doing that before on my previous configuration |
| 21:10.37 | bn43 | had it on the whole day |
| 21:11.51 | bn43 | maybe I bumped it on the way home but i doubt that would have made a difference - these phones are pretty rugged |
| 21:12.14 | *** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk) |
| 21:13.39 | Linuturk | I'm trying to find a solution that works as follows. I want to be able to allow users to use a regular analog phone at their homes, but have an ATA register to my asterisk server and allow them to make calls to other user's, all without messing with the user's gateway firewalls |
| 21:14.03 | Linuturk | maybe an ATA with an OpenVPN client? |
| 21:14.33 | bn43 | another thing that worked on my previous configuration was that I got a vmail notification on the phone and received the email - now no vmail message - I have followed this http://www.voip-info.org/wiki/view/Asterisk+sip+mailbox but no luck |
| 21:16.19 | lilkid | a2bill vs astbill, anyones opinions? deciding whether to ditch astbill for a2bill |
| 21:19.18 | *** join/#asterisk adam000 (n=adam@c-76-97-76-93.hsd1.ga.comcast.net) |
| 21:22.55 | rwaite | astbill killed my first born child, i'd stay away. |
| 21:23.26 | Micc | Is there a simple way to impliment hunt groups without that agi php script? |
| 21:23.59 | Micc | I know its simple to get the channel state and move to next if in use. But I just need a little example dialplan code. |
| 21:24.27 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
| 21:25.31 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek) |
| 21:27.50 | [TK]D-Fender | checkout time, later all |
| 21:28.11 | bn43 | anyone know how I can get the mwi button to work on my snom phone? |
| 21:28.25 | bn43 | mwi indicator that is |
| 21:33.47 | bn43 | anyone? |
| 21:38.10 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 21:41.10 | rootforce | did you set mailbox in sip.conf |
| 21:41.21 | rootforce | bn43: ? |
| 21:41.29 | bn43 | yeah |
| 21:41.35 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:41.45 | *** join/#asterisk flohack (n=fhackenb@lancelot.acoveo.com) |
| 21:42.03 | flohack | Hi! Is someone familiar with SIP transfers on asterisk? |
| 21:42.09 | bn43 | got answer? :-) |
| 21:42.21 | *** join/#asterisk stevetotaro (n=Steve@pool-71-254-231-87.hrbgpa.east.verizon.net) |
| 21:42.21 | bn43 | its driving me nuts! |
| 21:43.18 | *** join/#asterisk SlicerDicer (n=kvirc@69-92-107-4.cpe.cableone.net) |
| 21:43.59 | lilkid | rwaite: hah, will do. |
| 21:45.36 | wonderworld | flohack: what do you want to do? |
| 21:45.39 | flohack | Does asterisk support direct attended transfers between sip users? Currently asterisk acknowledges the attended transfer and simply goes into the dialplan, event though there is already an active call the the transfer target . |
| 21:45.44 | *** join/#asterisk UQlev (n=kvirc@91.184.220.73) |
| 21:46.35 | bn43 | rootforce: sorry did not see your previous question |
| 21:46.42 | flohack | wonderworld: Hi! Ok I have three sip phones. A calls B, B wants to transfer A to C, B calls C, C picks up. B says: attended transfer (SIP REFER). Now the call from B->C should be replaced by the call from A->B |
| 21:46.50 | flohack | but that's not what happens. |
| 21:47.10 | bn43 | where do you set mailbox in sip.conf? is it the one u specify for each user? |
| 21:47.15 | flohack | as soon as I send SIP REFER, asterisk says OK and dials C using the dialplan |
| 21:47.38 | *** join/#asterisk brunner (n=chris@66.35.172.123) |
| 21:47.48 | rootforce | bn43: one moment i will give you a pastebin example |
| 21:47.49 | brunner | what is the best channel for asking about non-asterisk small business telephone systems? |
| 21:49.06 | wonderworld | flohack: are you sure you tested it with the right digits? did you check features.conf? |
| 21:49.09 | flohack | wonderworld: Here is the sip debug log http://pastebin.com/m758d1754 Thanks for helping me! |
| 21:49.18 | *** part/#asterisk mikealeonetti (n=mikel@static-72-68-153-122.nycmny.fios.verizon.net) |
| 21:49.31 | wonderworld | there is blind transfer and attended transfer |
| 21:49.37 | flohack | wonderworld: I don't want to use the features.conf transfer feature, because I'm using sip phones. |
| 21:49.51 | wonderworld | you can use it with sip phones |
| 21:50.03 | rootforce | bn43: http://pastebin.com/d64125b6f |
| 21:50.14 | flohack | wonderworld: I know, but SIP REFER is just so much better. |
| 21:50.24 | flohack | wonderworld: It used to work a while ago... |
| 21:50.26 | wonderworld | never did that, sorry |
| 21:50.43 | flohack | wonderworld: thanks a lot anyway! |
| 21:50.59 | flohack | Has someone else ever done SIP REFER attended transfer with asterisk? |
| 21:51.15 | bn43 | rootforce: is the vmbox their individual box? |
| 21:51.16 | rootforce | bn43: that goes in sip.conf in case i did not already mention that |
| 21:51.40 | rootforce | bn43: it is whatever mailbox you want to receive the notifications for |
| 21:51.56 | rootforce | bn43: 9 times out of 10 it will be their personal mailbox |
| 21:52.21 | bn43 | I followed http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ which shows (username)@default as an example |
| 21:52.54 | rootforce | bn43: there may be some additional configuration on the snom side. I have never used snom phones before, but what I sent you will tell asterisk what to send to the phone. |
| 21:52.56 | *** join/#asterisk jazzsunn (n=jaytown@COX-66-210-184-98-static.coxinet.net) |
| 21:53.04 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:55.27 | *** part/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 21:56.14 | bn43 | hey its now working!!! |
| 21:57.34 | bn43 | last one for the nite - when I access the mailbox I get the prompt saying "comedian mail - mailbox" and I have to wait about 3 seconds for password prompt |
| 21:57.53 | bn43 | how do I get to the password prompt immediately? |
| 21:57.53 | rootforce | bn43: press pound |
| 21:58.02 | rootforce | bn43: er # |
| 21:58.25 | bn43 | is there a way to take out the first bit? |
| 21:59.32 | rootforce | bn43: you can set up the dialplan so that when you dial your extensin from your extension you go straight to your vmbox |
| 22:01.08 | bn43 | According to the tutorial, I set the number to 4242 |
| 22:01.13 | bn43 | is that the problem? |
| 22:01.35 | citywok | has anybody else run into problems with SOX not accepting files named like asterisk-1251231.12512.wav |
| 22:01.48 | citywok | complaining about .12512 being an invalid extension |
| 22:02.12 | frogonwheels | citywok: that sounds right. you probably need to explicitly tell it the file format :| |
| 22:02.19 | rootforce | bn43: what tutorial are you using? |
| 22:02.29 | bn43 | http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 22:02.42 | citywok | frogonwheels: it works fine as long as the .12512 isn't there, its not smart enough to be able to handle files with .'s in the name |
| 22:02.48 | bn43 | recommended by the guru's here :-) |
| 22:03.00 | frogonwheels | citywok: I actually ran into that error yesterday :0 |
| 22:03.23 | citywok | hahaha, may i ask what the option is to specify type then, if you just played this game hahaha |
| 22:03.49 | frogonwheels | citywok: sox -t wav |
| 22:03.55 | frogonwheels | citywok: I think |
| 22:04.21 | rootforce | bn43: try dialing extension 100 from the phone that is registered to 100 and press * |
| 22:04.31 | Linuturk | can someone recommend a good analog telephone adapter with VPN capabilities? |
| 22:04.51 | citywok | frogonwheels: yup, i just found it on google too, tyvm, that was an easy fix |
| 22:05.15 | bn43 | it rings itself |
| 22:05.16 | citywok | it was already specified on my sox command, but not on the soxmix, whoops |
| 22:05.16 | frogonwheels | citywok: you could try sox -t ${FNAME#*.} ${FNAME} outputfilename |
| 22:05.43 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 22:05.44 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 22:05.48 | frogonwheels | np citywok |
| 22:05.59 | *** join/#asterisk fogo (n=Paul@69.169.132.35) |
| 22:08.42 | lilkid | does a2billing manage the configuration of asterisk? e.g. accounts/dialplan etc., or does it just handle billing? |
| 22:10.08 | manxpower | AEL2 sure does make complex dialplans easier to read |
| 22:10.30 | citywok | i have one other problem. i'm receing a call from sip, and then dialing out of a zap channel, but the callid seems to be incremented by one, any ideas? dialplan: http://pastebin.com/d1a6b78f |
| 22:10.49 | rootforce | bn43: hmm simple fix is to change [stations] to the following http://pastebin.com/m17301829 |
| 22:11.28 | citywok | crap, nevermind, i know what it is |
| 22:11.51 | bn43 | do I replace the top 2 with the bottom 2? |
| 22:12.57 | [TK]D-Fender | lilkid: Just billing with some minimal dialplan you have to insert |
| 22:13.36 | bn43 | oh stupid! sorry |
| 22:13.48 | bn43 | its early in the morning here |
| 22:14.54 | Micc | I think asterisk-gui converted my voicemail.conf to mac format. |
| 22:15.01 | Micc | How do I convert it back to unix format? |
| 22:16.30 | bn43 | thank you! |
| 22:17.20 | [TK]D-Fender | Micc: what is "mac format", and why do you think that? |
| 22:17.52 | Micc | TKD-Fender, because it has ^M at the end of each line instead of a line feed. |
| 22:18.01 | Micc | And pico says its mac format. |
| 22:18.02 | [TK]D-Fender | Micc: tahts actually DOS format... |
| 22:18.03 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
| 22:18.11 | [TK]D-Fender | Micc: "dos2unix thefile" |
| 22:18.20 | Micc | TKD-Fender, dos is actually CRLF. |
| 22:20.08 | flohack | wonderworld: I found the problem. It's a bug in asterisk. It searches for a "REPLACES=" in the Refer-To header, but only at the beginning. My softphone sends: Require=replaces&Replaces=9ccf and therefore asterisk does not detect an attended transfer, but a blind transfer.... |
| 22:20.21 | Micc | dos2unix isn't working. |
| 22:20.31 | Micc | It says it converted but the file is exactly the same. |
| 22:20.50 | rue_mohr | didyou open it in windows? |
| 22:20.50 | frogonwheels | Am I correct in thinking that the only way of 'background'ing SayUnixTime for inside an IVR is through the agi system? |
| 22:21.16 | rue_mohr | Micc, it just changes the codes used for end of line |
| 22:21.28 | Micc | Ok, I had to tell it what type of file it was. |
| 22:21.44 | Micc | dos2unix filename doesn't work but dos2unix -c Mac filename does work. |
| 22:22.06 | rue_mohr | Micc, if you use hd (iirc) you can see the old linefeeds and the new one |
| 22:22.09 | frogonwheels | [TK]D-Fender: just ^M is mac. (I know from vim) |
| 22:22.40 | frogonwheels | [TK]D-Fender: if you see ^M at the end of your lines - you're reading DOS format as unix. |
| 22:23.03 | [TK]D-Fender | I'm sure there is a similar easy script to correct.... |
| 22:23.23 | Micc | Its good now. |
| 22:23.39 | Micc | asterisk gave me a strange error though. it said line 2 did not contain an = |
| 22:24.02 | frogonwheels | Micc: vim's always good for changing different file-formats too :) |
| 22:24.17 | *** join/#asterisk ftp3 (n=none@pool-71-117-187-57.ptldor.dsl-w.verizon.net) |
| 22:24.18 | manxpower | does anyone know in AEL2 if case ${MYVAR} < 100: is valid? |
| 22:24.44 | Micc | vim seems like too much learning curve. |
| 22:24.46 | ftp3 | hi, question.. in a billing rates .csv.. is "1,1" the same as "60,60" ? |
| 22:25.02 | daniev | hello guys. i'm selling an sangoma A101D card. never used |
| 22:25.07 | frogonwheels | Micc: not too much. if you run vimtutor you can get going quickly. |
| 22:25.23 | [TK]D-Fender | manxpower: Boolean case = if :) |
| 22:25.24 | Micc | I know basic vi commands. |
| 22:25.49 | frogonwheels | Micc: well vimtutor is still worth running. it shows some of the basic vim extensions as well. |
| 22:25.55 | flohack | Is an asterisk core hacker here? I'd like to have a chat about a patch to chan_sip.c |
| 22:26.12 | frogonwheels | Micc: and vim has excellent documentation. |
| 22:26.19 | *** join/#asterisk ghenry (n=ghenry@92.41.230.114.sub.mbb.three.co.uk) |
| 22:26.28 | *** join/#asterisk myselfhimself (n=jonathan@ip-33.net-82-216-240.rev.numericable.fr) |
| 22:26.30 | myselfhimself | hi |
| 22:26.35 | manxpower | flohack: dev questions should be on #asterisk-dev |
| 22:26.54 | flohack | manxpower: Oh, ups...sorry... |
| 22:26.59 | myselfhimself | I really need help to configure a simple softphone+asterisk on a same local host for SIP |
| 22:27.28 | myselfhimself | I have tutorials before my eyes but I really can't manage to make it work even when following those tutorials precisely |
| 22:27.44 | myselfhimself | for example if someone would want to help me out with ekiga... |
| 22:27.46 | [TK]D-Fender | myselfhimself: Softphon on the same PC as *? |
| 22:27.46 | frogonwheels | myselfhimself: look at the errors generated on the asterisk -r console |
| 22:28.14 | myselfhimself | [TK]D-Fender yes |
| 22:28.20 | myselfhimself | frogonwheels there's no error for now |
| 22:28.26 | myselfhimself | and I do have loaded the sip module |
| 22:28.32 | myselfhimself | with module load chan_sip.so |
| 22:28.45 | frogonwheels | myselfhimself: so you're seeing it register? or nothing? |
| 22:28.52 | myselfhimself | upon each change of sip.conf, I do sip reload |
| 22:28.52 | [TK]D-Fender | myselfhimself: in your peer setp "port=5061" and change the port your softphone binds to to 5061 |
| 22:28.57 | myselfhimself | I see nothing registering |
| 22:29.09 | [TK]D-Fender | myselfhimself: otherwise * and your softphone will fight over the port and bad things will happen |
| 22:29.37 | myselfhimself | [TK]D-Fender what do you mean by "in your peer setup", sip.conf ? |
| 22:29.55 | [TK]D-Fender | myselfhimself: Yes |
| 22:30.09 | myselfhimself | I had that port change thing in a tutorial |
| 22:30.22 | myselfhimself | I'm putting that back doing sip reload |
| 22:31.16 | [TK]D-Fender | myselfhimself: also do "sip show peers" to make sure that the module is even loaded. Then do "sip show peer [yourpeer]" after attempting to register to confirm if it has. All the while make sure to have enabled gloabl SIP DEBUG. |
| 22:31.29 | myselfhimself | hey |
| 22:31.40 | [TK]D-Fender | global |
| 22:31.44 | myselfhimself | ok for the debug and sip show peers |
| 22:31.59 | myselfhimself | ekiga has waken up and has tried to login |
| 22:32.05 | myselfhimself | so I sse things in the asterisk CLI |
| 22:33.20 | myselfhimself | I see thinks like Trying, Non Authorized and then Destruction blocks in the CLI |
| 22:33.22 | brunner | can asterisk work with existing Nortel Norstar phones? |
| 22:33.59 | [TK]D-Fender | myselfhimself: Progress... they're talking.. and DISAGREEING. Excellent |
| 22:33.59 | rob0 | Nope, only with phones that do NOT exist. |
| 22:34.40 | [TK]D-Fender | brunner: There are gateway devices for those, but the words "very not cost effective" come to mind |
| 22:34.46 | rob0 | iPhone = Phone * i |
| 22:34.59 | brunner | [TK]D-Fender: what sort of gateway device? |
| 22:35.47 | myselfhimself | what is the qualify= option for ? |
| 22:37.02 | [TK]D-Fender | brunner: http://www.addvant.com/index.php?main_page=product_info&products_id=492 |
| 22:37.23 | [TK]D-Fender | myselfhimself: NAT keepalive primarily |
| 22:37.34 | [TK]D-Fender | myselfhimself: if thats what its complaining about, don't worry |
| 22:37.43 | [TK]D-Fender | myselfhimself: Watch the regsiter & invite requests |
| 22:38.00 | myselfhimself | ok |
| 22:39.38 | brunner | if I wanted to use an asterisk box to take advantage of unused channels on my T1, would it be possible while the current system is still set up? |
| 22:39.45 | brunner | I have a nortel system right now |
| 22:39.56 | brunner | one of these: http://www.craigcommunications.net/norstar-modular-ics-0x32-cabinet-nt7b53fa-93.asp |
| 22:40.32 | [TK]D-Fender | brunner: Clarify "take advantage" |
| 22:40.40 | brunner | use them to make outbound calls |
| 22:40.58 | [TK]D-Fender | brunner: Why are they "unused" right now? |
| 22:41.14 | *** join/#asterisk saftsack (n=saftsack@p57924D3A.dip.t-dialin.net) |
| 22:41.18 | brunner | [TK]D-Fender: because I'm not taking calls for an internet radio station right now |
| 22:41.28 | *** join/#asterisk keebler (i=9446c2d5@gateway/web/ajax/mibbit.com/x-52279784c00b11c5) |
| 22:41.52 | [TK]D-Fender | brunner: If you want * inline with your existing PBX its certainaly doable with a 2-port card. |
| 22:42.00 | keebler | Where's the best play to buy VOIP hardware? |
| 22:42.06 | keebler | buy |
| 22:42.08 | keebler | place |
| 22:42.19 | keebler | Damn brain. |
| 22:42.31 | keebler | Where is the best place to purchase VOIP hardware? |
| 22:42.44 | [TK]D-Fender | keebler: in USA : www.telephonydepot.com is very competitive and very good service |
| 22:42.49 | keebler | And wireless bridges. |
| 22:43.14 | brunner | [TK]D-Fender: how would that work? |
| 22:43.55 | [TK]D-Fender | brunner: telco got into *, PBX goes into *. * takes calls on the channels you want and dials them out to the norstar and does what it wants with the rest |
| 22:44.51 | keebler | Holy crap [TK]D-Fender, they have some damn good prices. |
| 22:45.33 | [TK]D-Fender | keebler: Probably the best place for general VoIP gear |
| 22:46.20 | keebler | yeah, just needed 8-ATAs. And a couple cheap phones. |
| 22:47.41 | keebler | Now I need to find a good online supplier for some bridges. :) |
| 22:52.55 | ftp3 | so, does anyone know... in a billing rates .csv.. is "1,1" the same as "60,60" ? |
| 22:53.05 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 22:56.39 | [TK]D-Fender | ftp3: Whose "billing rates" CSV? |
| 22:57.10 | myselfhimself | [TK]D-Fender I think that I neeed to set a realm for the connection to work |
| 22:57.17 | *** join/#asterisk Bonix (n=Bonix@212-lo1.rt2.isimples.com.br) |
| 22:57.18 | rootforce | keebler: i agree with [TK]D-Fender td is very good |
| 22:57.20 | myselfhimself | my Ekiga version doesn't seem to allow that |
| 22:57.32 | myselfhimself | I'll get a newer one |
| 22:57.55 | brunner | [TK]D-Fender: what equipment would I need for that? it's not a PRI line. it's a channelized T1. |
| 22:58.19 | rootforce | keebler: what kind of bridges are you looking for? |
| 22:59.08 | [TK]D-Fender | brunner: 2 port digital card |
| 22:59.52 | *** join/#asterisk zamba (i=marius@sveigde.hih.no) |
| 22:59.56 | keebler | rootforce: , Outdoor bridges. |
| 23:00.13 | zamba | what's the closest thing we have to a bullet-proof nat setup? |
| 23:00.33 | [TK]D-Fender | keebler: Suspension, or straight support beams? |
| 23:00.35 | zamba | i need a howto or document that describes this and all the possible failures that can occur when trying to set up a connection |
| 23:00.42 | [TK]D-Fender | ~sipnat |
| 23:00.43 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 23:00.44 | [TK]D-Fender | ^^^^^^^^^^^ |
| 23:00.51 | keebler | rootforce: Right now I'm using WRT54Gs, but they have MAC address issues. And if I'm going to be deploying 230 of these things, I don't want to manually have to change each one. |
| 23:01.02 | zamba | [TK]D-Fender: thanks |
| 23:01.04 | rootforce | keebler: 802.11 or something good |
| 23:01.17 | keebler | huh? |
| 23:01.27 | keebler | 902.11 |
| 23:01.31 | keebler | emr |
| 23:01.32 | keebler | 802.11 |
| 23:01.35 | *** join/#asterisk CapRiCoRN^80 (n=carp@c80-216-221-198.bredband.comhem.se) |
| 23:01.36 | johnakabean | hey room, any way to disable "please leave your message after the tone" for just one extensions using custom context or similiar? |
| 23:01.38 | CapRiCoRN^80 | hi all |
| 23:01.38 | keebler | I only need to go 950ft. |
| 23:01.42 | [TK]D-Fender | 902.10? |
| 23:01.47 | zamba | [TK]D-Fender: if you set nat=yes globally, how will this work if the asterisk server isn't behind nat? |
| 23:01.57 | johnakabean | no effect zamba |
| 23:02.00 | zamba | [TK]D-Fender: i've set this option, believing this to then be a default setting for all peers..? |
| 23:02.03 | [TK]D-Fender | johnakabean: "s" <- |
| 23:02.04 | zamba | ah, ok |
| 23:02.12 | zamba | johnakabean: thanks :) |
| 23:02.20 | ftp3 | d-fender, i was just googleing peoples rates.. and i see some people say 1,1 and some people say 60,60 (and other variables), but I think 1,1 and 60,60 are the same.. but I am not sure.. so I was asking |
| 23:02.28 | [TK]D-Fender | zamba: Set per each |
| 23:02.31 | johnakabean | it does have an effect of course if you do have a nat, for nat = no |
| 23:02.41 | zamba | [TK]D-Fender: hm? |
| 23:02.50 | [TK]D-Fender | zamba: It can have a nasty impact on your ITSP's, not so much for your remote phones. |
| 23:02.50 | zamba | [TK]D-Fender: i should set nat=yes for every client? |
| 23:02.57 | [TK]D-Fender | zamba: Yes |
| 23:02.59 | zamba | ITSP? |
| 23:03.03 | [TK]D-Fender | ~itsp |
| 23:03.04 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 23:03.14 | rootforce | keebler: you might look at http://www.streakwave.com/index.asp or http://store.wisp-router.com/index.asp |
| 23:03.23 | [TK]D-Fender | zamba: This is in the guide... |
| 23:04.10 | zamba | i've set all the peers to nat=yes now, but when i do 'sip show peers' it says 'N' on every user for the nat column.. hm.. |
| 23:04.14 | johnakabean | [TK]D-Fender: VM Context: default,s ?? |
| 23:04.33 | [TK]D-Fender | johnakabean: voicemail CLI OPTION |
| 23:04.44 | johnakabean | i understand the dialplan to how to add it |
| 23:04.46 | johnakabean | but in freepbx |
| 23:04.50 | johnakabean | is my question |
| 23:04.55 | [TK]D-Fender | johnakabean: Bend over, insert shaft |
| 23:04.59 | rootforce | keebler: you are deploying 230 bridges? |
| 23:05.13 | johnakabean | if i edit the dialplan freepbx will overwrite it |
| 23:05.29 | [TK]D-Fender | johnakabean: If they don't offer it on the "Extension setup" then you're FUBAR'd |
| 23:05.38 | [TK]D-Fender | johnakabean: See above... |
| 23:06.28 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 23:06.51 | keebler | ahah rootforce, YES, But not all on the same network. |
| 23:07.20 | keebler | 30 separate networks. |
| 23:07.25 | keebler | 8 bridges each. |
| 23:07.51 | keebler | And behind each bridge is a Asterisk phone. |
| 23:08.37 | brunner | [TK]D-Fender: does digium sell the kind of 2-port card that I would need? |
| 23:09.14 | johnakabean | attach=no|saycid=yes|envelope=yes|delete=no anyway to add it here |
| 23:09.16 | johnakabean | ? |
| 23:09.34 | myselfhimself | hey |
| 23:09.43 | myselfhimself | with kphone I manage to connect to my asterisk server |
| 23:09.45 | myselfhimself | though I get Looking for 007 in home (domain 127.0.0.1) |
| 23:09.45 | myselfhimself | Disconnected from Asterisk server |
| 23:09.51 | myselfhimself | from the asterisk windows |
| 23:10.05 | myselfhimself | and asterisk doesn't list in the processes anymore (so it's shutdown) |
| 23:10.27 | myselfhimself | and that happened when I tried to call 007@localhost from kphone (it had been able to register first) |
| 23:10.36 | [TK]D-Fender | brunner: All the major makers do |
| 23:10.49 | myselfhimself | so .. the reason for that is that my extensions.conf is misformatted and makes asterisk crash ? |
| 23:11.07 | brunner | [TK]D-Fender: so any of these T1 cards are capable of acting like the C/O does, as far as my existing Nortel system is concerned? |
| 23:11.11 | bmoraca | keebler: no luck with those EZGO bridges? |
| 23:11.20 | [TK]D-Fender | brunner: All of them |
| 23:11.25 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:11.35 | keebler | bmoraca: I can't find ANY reviews on them. so I'm very scared. |
| 23:11.50 | bmoraca | well, best you can do is try them, really |
| 23:12.02 | keebler | I'm buying all the equipment today though... And just wanted to avoid wasting time. |
| 23:12.13 | keebler | bmoraca: I've got 7 days. |
| 23:12.16 | bmoraca | you're buying it ALL? without testing them? |
| 23:12.21 | bmoraca | ouch |
| 23:12.49 | CapRiCoRN^80 | [TK]D-Fender: http://pastebin.com/m4169736d |
| 23:13.17 | CapRiCoRN^80 | i have read some tutorials for NAT and come up with follwing linex |
| 23:13.19 | CapRiCoRN^80 | lines |
| 23:13.38 | keebler | bmoraca: No. I'm buying 8. |
| 23:13.43 | bmoraca | oh, ok |
| 23:13.44 | bmoraca | lol |
| 23:13.48 | keebler | bmoraca: enough for one system. |
| 23:14.06 | [TK]D-Fender | CapRiCoRN^80: Read AGAIN : |
| 23:14.08 | [TK]D-Fender | ~sipnat |
| 23:14.09 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 23:14.23 | vader-- | does the cisco ata 186 take the same config files and use the same tftp loading process as the 7940g phones? |
| 23:14.26 | CapRiCoRN^80 | ok |
| 23:14.34 | bmoraca | keebler: remember that they're directional with a 35 degree arc...you may not be able to get all remote clients with a single central access point...and they may not support point to multipoint |
| 23:14.38 | johnakabean | oh sit fender, i forgot, i could create a voicemail_custom.conf and then include it and freepbx won't delete that line. |
| 23:15.05 | johnakabean | * the include line |
| 23:15.20 | keebler | bmoraca: Well, I do know they support upto 6 in WDS mode. |
| 23:15.30 | keebler | bmoraca: I forgot the rest of the stats, |
| 23:15.38 | bmoraca | WDS is no good for VoIP... |
| 23:15.47 | keebler | For the ones that CAN't get directly, I was going to put some Omni's. |
| 23:16.01 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:16.16 | keebler | Wait... My gateway isn't going to be an EZgo, so if they're all the Multipoints pointing to the POINT. |
| 23:16.16 | bmoraca | at BEST you'll be getting 3mbps throughput, and that's assuming there's enough power to push a full 54mbit |
| 23:16.20 | keebler | Should it matter? |
| 23:16.36 | bmoraca | what's your central access point going to be? |
| 23:16.37 | keebler | I wasn't goint to use WDS, jsut saying thats all I remember of the stats. |
| 23:16.41 | bmoraca | oh, ok |
| 23:17.27 | brunner | Right now my organization has 20 extensions, each with its own dedicated T1 channel. Each channel has its own phone number. Would it be possible to use only 10 of those T1 channels to service 20 extensions, and still keep the 20 phone numbers assigned to their respective extensions? |
| 23:17.34 | keebler | bmoraca: Well, for the time being, a Basic DDWRT router. |
| 23:21.29 | brunner | are PRI lines significantly more expensive than channelized T1s? |
| 23:22.47 | [TK]D-Fender | brunner: no reason for them to be |
| 23:22.56 | brunner | we currently have two channelized T1s. If I switch to PRI, can I use unused channels for internet service? |
| 23:23.03 | [TK]D-Fender | brunner: and you do whatever the hell you want with your channels. |
| 23:23.22 | [TK]D-Fender | brunner: T1 can be split voice / dart /CAS or PRI, your choice |
| 23:23.26 | [TK]D-Fender | data* |
| 23:24.04 | brunner | [TK]D-Fender: can the channels be dynamically allocated for different purposes? |
| 23:24.34 | [TK]D-Fender | brunner: Statically yes, dynamically no |
| 23:24.35 | brunner | I mean, if I'm using one PRI line for voice and internet, can I drop channels from the internet if my voice channels start getting used up? |
| 23:24.44 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 23:24.44 | *** mode/#asterisk [+o russellb] by ChanServ |
| 23:25.45 | *** join/#asterisk seanmh (n=johndoe@216.31.95.99) |
| 23:26.56 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:27.08 | Mark17 | brunner: technically: yes |
| 23:27.37 | brunner | Mark17: is it like almost impossible to setup that way? |
| 23:28.20 | *** part/#asterisk ftp3 (n=none@pool-71-117-187-57.ptldor.dsl-w.verizon.net) |
| 23:30.29 | johnakabean | [TK]D-Fender: 202@default,s is allowed in the extensions setup of freepbx but it ignores it. |
| 23:30.30 | Mark17 | if i understand it correctly it would take some time, but it is possible |
| 23:30.43 | brunner | if we switched to PRI, we could stop paying for 24 channels if we never use more than ten, right? |
| 23:31.02 | johnakabean | i even tried the | sign |
| 23:31.14 | Mark17 | depends on how the design is currently |
| 23:31.18 | *** join/#asterisk edibrac (n=elusive4@206.173.193.34.ptr.us.xo.net) |
| 23:31.24 | drmessano | [TK]D-Fender: Since when do you support FreePBX ??? :P |
| 23:31.43 | johnakabean | ok, drmessano, you're the freepbx expert |
| 23:31.48 | edibrac | what's the going rate for an asterisk consultant? |
| 23:31.54 | edibrac | $150-200/hour? |
| 23:32.05 | brunner | edibrac: I was quoted 170 today |
| 23:32.10 | johnakabean | i'm trying to disable the vm-intro for one extension but if i add it in the dialplan, of course asterisk will overwrite it |
| 23:32.21 | [TK]D-Fender | johnakabean: if freePBX ignores it, TFB. * ignoring it would be another matter |
| 23:32.26 | brunner | edibrac: sorry, it was 175 |
| 23:32.38 | *** part/#asterisk digitalirony (i=digitali@my.grandma.uses.shellium.org) |
| 23:32.48 | johnakabean | freepbx will overwrite it |
| 23:32.55 | Mark17 | i normally ask 60 E/hour, but for persons with papers and that know asterisk totally 200 sounds normall |
| 23:32.58 | *** join/#asterisk digitalirony (i=digitali@my.grandma.uses.shellium.org) |
| 23:33.02 | [TK]D-Fender | brunner: PRI is just a T1 signalling. How many channel you want is still your choice |
| 23:33.23 | *** join/#asterisk BadHAL (n=wut@cpe-72-179-194-139.stx.res.rr.com) |
| 23:33.26 | johnakabean | tfb? |
| 23:33.31 | [TK]D-Fender | ~tfb |
| 23:33.31 | jbot | tfb is, like, Too #&^$ing bad.... |
| 23:33.39 | [TK]D-Fender | :D |
| 23:33.58 | johnakabean | ~ftfb |
| 23:34.11 | brunner | god, I hate Nortel. |
| 23:34.13 | johnakabean | where's jbot for that one fender |
| 23:35.05 | johnakabean | anyway, drmessano, i have tried adding it to VM context too |
| 23:35.23 | brunner | Does this mean I don't need a physical PRI card for my Nortel system? http://www.craigcommunications.net/norstar-pri-key-code-ntab2769.asp |
| 23:36.02 | edibrac | can anyone here recommend a good asterisk consultant in the bay area? |
| 23:36.07 | edibrac | San Francisco bay area |
| 23:36.41 | [TK]D-Fender | johnakabean: this is a dialplan option, not a BOX option. |
| 23:37.06 | edibrac | the problem is this - HDLC errors from different hardware setups (2 supermicro boxes, and an ASUS mobo, with 3 different Digium cards) |
| 23:37.17 | [TK]D-Fender | brunner: Depends what signalling you want to sue with it. |
| 23:37.21 | Mark17 | is it possible to include a file in the extensions.conf? so it would be used like all content from that file was located in extensions.conf (with the context where you include it)? |
| 23:37.27 | drmessano | johnakabean: You're wasting your time. Not helping you. I could be the bigger man and ignore how you've been a total dick towards me when i've helped you, then boasted about it, but I would rather be the bigger douche. |
| 23:37.37 | drmessano | So GLWT |
| 23:38.14 | [TK]D-Fender | use* |
| 23:38.21 | jaytee | brunner, you hate Nortel? wow! what a surprise :-) |
| 23:38.43 | brunner | "The Norstar PRI Key Code is NOT a generic key code. The Norstar PRI Key Code requires a Norstar certified installer to program the system." |
| 23:39.31 | jaytee | hmmm, and they're filing for bankruptcy protection in the US because...??? |
| 23:39.32 | drmessano | brunner: Try 1111 |
| 23:39.34 | drmessano | heh |
| 23:39.34 | johnakabean | sorry, messano, that' your naive to think i have been a dick. With every time i have tried to help someone the best i can, I get smart ass, demeaning remarks from you; its like a hypocritical think with what you say. |
| 23:39.43 | johnakabean | and I don't believe you know how to do it. |
| 23:40.07 | drmessano | johnakabean: If you didnt believe I could, you wouldnt have asked. Now you're showing just why I wont help you. |
| 23:40.08 | *** join/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec) |
| 23:40.17 | drmessano | johnakabean: So good luck |
| 23:40.37 | johnakabean | NO, i'm proving the point you have no wish to help someone you're afraid of |
| 23:41.03 | drmessano | I'm afraid of you? |
| 23:41.08 | johnakabean | you think by me trying to help someone and suggest something, and if it worked, you would lose reputation in here |
| 23:41.08 | drmessano | I have chunks of you in my stool |
| 23:41.11 | johnakabean | you think its a game |
| 23:41.15 | drmessano | LOL |
| 23:41.19 | drmessano | Of course |
| 23:41.34 | johnakabean | play your game and I'll live a life |
| 23:41.38 | *** join/#asterisk kisu (n=kisu@2001:5c0:1100:9900:1d37:23b5:94db:ec35) [NETSPLIT VICTIM] |
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| 23:41.45 | drmessano | Yes sir |
| 23:41.46 | drmessano | You found me out |
| 23:42.03 | drmessano | borrows jaytee mascara and puts on some yellowcard |
| 23:42.29 | jaytee | that color just doesn't go with your skin tone! |
| 23:42.30 | johnakabean | i have said nothing off the wall to you and you come out with a hypocritical remark that I'm the dick? |
| 23:42.34 | johnakabean | whatever |
| 23:42.41 | drmessano | jaytee: :( |
| 23:42.48 | *** part/#asterisk amessina (n=amessina@2001:470:1f11:68:20e:cff:fe01:d5ec) |
| 23:43.15 | johnakabean | Oh, and 1.6 is not a drop in for everyone; a microsoft remark. |
| 23:43.24 | drmessano | Actually it is |
| 23:43.31 | drmessano | Just not for you |
| 23:43.36 | drmessano | Since you cant do it |
| 23:43.42 | drmessano | and seems you cant do a lot.. |
| 23:43.44 | drmessano | :( |
| 23:43.54 | johnakabean | sorry i followed the guidelines for my provider which wouldn't work on 1.6 |
| 23:44.00 | johnakabean | fender was there and verified |
| 23:44.04 | drmessano | :( |
| 23:44.10 | jaytee | 1.6? other than some SIP TCP issues it's a walk in the park! |
| 23:44.16 | drmessano | Yeah, I dont buy that |
| 23:44.35 | johnakabean | i'm not here to prove anything to you; you prove you're a dick on your own |
| 23:44.36 | manxpower | You can learn about the 1.6 changes in upgrade.txt |
| 23:44.39 | *** join/#asterisk kisu (n=kisu@2001:5c0:1100:9900:1d37:23b5:94db:ec35) [NETSPLIT VICTIM] |
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| 23:44.58 | johnakabean | People don't gravel at your feet messano |
| 23:45.09 | drmessano | johnakabean: :) |
| 23:45.19 | drmessano | jaytee: 18 or 25? |
| 23:45.41 | *** join/#asterisk tobias (n=tobias@user-0ce2hu8.cable.mindspring.com) |
| 23:45.48 | [TK]D-Fender | plenty of gravel at drmessano's feet. Now THERE's a foundation we can build on! |
| 23:46.13 | [TK]D-Fender | johnakabean: and as I've said, this is not s VM box parm, its the CLI call. |
| 23:46.19 | jaytee | 18 or 25? what? |
| 23:47.13 | johnakabean | excuse me grovell |
| 23:47.19 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 23:47.25 | jaytee | one l in grovel |
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| 23:48.33 | johnakabean | I know, fender, but there is an input field in freepbx to edit it |
| 23:48.45 | brunner | Strom Carlson is such a dick |
| 23:48.58 | johnakabean | messano, is your last name carlson? |
| 23:49.22 | drmessano | johnakabean: Keep proving me right |
| 23:49.42 | johnakabean | whenever i put comma S (,s) behind the other parameters it puts in the dialplan, it doesn't get carried over |
| 23:49.58 | [TK]D-Fender | johnakabean: Ok, so show me what the field looks like (imagbin) and what it does (pastebin the CLI) |
| 23:50.18 | brunner | hey Corydon76-dig, very long time no chat |
| 23:50.33 | [TK]D-Fender | brunner: Why do you say that? |
| 23:50.44 | brunner | [TK]D-Fender: because he was rude to me |
| 23:50.55 | [TK]D-Fender | brunner: Where? concerning what? |
| 23:50.57 | Corydon76-dig | brunner: greetings |
| 23:51.00 | CapRiCoRN^80 | [TK]D-Fender: http://pastebin.com/m3ba48ad |
| 23:51.08 | CapRiCoRN^80 | [TK]D-Fender: check now please |
| 23:51.22 | brunner | AIM, concerning Nortel and vendor lock-in |
| 23:51.29 | jaytee | hurry! and don't give me any shit!!! |
| 23:51.51 | [TK]D-Fender | CapRiCoRN^80: NAT stuff looks better, your dilaplan is broken however |
| 23:51.52 | brunner | [TK]D-Fender: I'm a friend of noogums -- I think that was his nick |
| 23:52.18 | Corydon76-dig | brunner: I'm not aware that n00gums has ever gotten involved in Asterisk |
| 23:52.33 | [TK]D-Fender | brunner: Well he's perfectly right about vendor lock in. You're already looking to expand a DEAD-END system |
| 23:52.34 | johnakabean | [TK]D-Fender: http://imagebin.org/36918 |
| 23:53.00 | brunner | [TK]D-Fender: no, I complained about the vendor lock-in, and he told me to grow up |
| 23:53.13 | [TK]D-Fender | johnakabean: that field looks more like you attempting to ABUSE it than it intending for you to add stuff to the end |
| 23:53.27 | [TK]D-Fender | brunner: .... GROW UP :) |
| 23:53.31 | brunner | har har |
| 23:53.34 | CapRiCoRN^80 | [TK]D-Fender: can you tell me how its broken ? |
| 23:53.37 | [TK]D-Fender | brunner: Its a dead end, so stop whining. |
| 23:53.42 | [TK]D-Fender | brunner: You can now add me to that list. |
| 23:53.51 | [TK]D-Fender | CapRiCoRN^80: "SIP," <- |
| 23:54.24 | brunner | Corydon76-dig: I don't think he has. I meant to address you, not [TK]D-Fender |
| 23:54.38 | jaytee | I've got 65% of my users migrated from Nortel to Asterisk. In another month and a half I'll be able to pull the plug on the Option 11C and put the bitch on Ebay. |
| 23:55.13 | Corydon76-dig | brunner: I'm aware of who you are |
| 23:55.15 | *** part/#asterisk Khratos (n=khratos@190.166.103.146) |
| 23:55.30 | Corydon76-dig | brunner: Names sometimes escape me, but I never forget a cutie |
| 23:55.44 | brunner | jaytee: how did you get the systems to inter-operate during the migration? |
| 23:55.53 | brunner | Corydon76-dig: apparently not =p |
| 23:56.06 | Corydon76-dig | brunner: PRI is going to be your best bet |
| 23:56.06 | jaytee | brunner, through a mixture of trickery and cunning |
| 23:56.22 | manxpower | jaytee: quite a bit of both, I imagine. |
| 23:56.28 | CapRiCoRN^80 | [TK]D-Fender: you mean the problem is in sip.conf ? . i wish you tell in straight words |
| 23:56.43 | jaytee | manxpower, ya don't know the half of it buddy :-) How's the new job treating ya? |
| 23:56.55 | manxpower | jaytee: so-so |
| 23:57.04 | [TK]D-Fender | CapRiCoRN^80: I said your DIALPLAN. |
| 23:57.05 | Corydon76-dig | The best method I've seen, to minimize the amount of tinkering you have to do with the Nortel is to put a passthrough on all of the lines going to the Nortel |
| 23:57.10 | CapRiCoRN^80 | ok |
| 23:57.16 | brunner | Corydon76-dig: yeah, I'm just trying to figure out what would be involved in switching the current system over to PRI |
| 23:57.21 | [TK]D-Fender | CapRiCoRN^80: then I quoted precise characters you could have TEXT SEARCHED. |
| 23:57.29 | manxpower | jaytee: I still have a former customer with a frankenpbx - Asterisk and Nortel MICS |
| 23:57.31 | Corydon76-dig | brunner: what is it currently, E&M? |
| 23:57.31 | jaytee | :-( oh, well, better than poverty or unemployement I guess is good enough for now in this economy. |
| 23:57.43 | Corydon76-dig | brunner: you can frontend the E&M, as well |
| 23:57.58 | manxpower | jaytee: I can't form a good opinion until I've been here for a few months |
| 23:58.11 | jaytee | manxpower, understandable |
| 23:58.19 | brunner | Corydon76-dig: it's one of these. what is the short hand for these models? http://www.craigcommunications.net/norstar-modular-ics-0x32-cabinet-nt7b53fa-93.asp |
| 23:58.35 | manxpower | jaytee: their customers are just as stupid as my customers were. they are just as disorganized as I am. |
| 23:59.00 | manxpower | brunner: MICS, IIRC |
| 23:59.09 | brunner | manxpower: thanks |
| 23:59.11 | CapRiCoRN^80 | [TK]D-Fender: brb |
| 23:59.17 | manxpower | and the MICS have about as much smarts as a turnip |
| 23:59.29 | jaytee | actually I've met smarter turnips |
| 23:59.30 | brunner | so I've learned |
| 23:59.48 | jaytee | but I never worked on a 0x32. just a 0x16 |
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