00:02.21 | SeRi | I have set to drop silently. |
00:02.37 | SeRi | usually hangs most scanners or attempt |
00:03.25 | p3nguin | As far as I know, pf only blocks or accepts. There's nothing in the middle. Of each of those, you can log or not log. |
00:04.39 | *** join/#asterisk luckman212_ (~do-not-re@static-108-46-165-162.nycmny.fios.verizon.net) |
00:05.01 | luckman212_ | anybody in here have any experience configuring Patton SmartNode gateways? |
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00:39.11 | cusco | hi... |
00:39.31 | cusco | p3nguin: im trying to test gtalk now, but calls from gtalk to asterisk are not comming in |
00:39.39 | cusco | context is set.. |
00:39.52 | cusco | I don't know what Im missing |
00:40.10 | cusco | jabber debug shows the call in |
00:48.08 | *** join/#asterisk snuff-work (~snuffy@210.9.82.197) |
00:48.39 | p3nguin | I'd have to see your jabber.conf, gtalk.conf, and the relevant context in extensions.conf. |
00:50.05 | cusco | ok hold |
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00:53.21 | cusco | p3nguin: http://paste.debian.net/136342/ |
00:54.43 | p3nguin | I see two things that stand out to me. |
00:55.25 | p3nguin | You specify a context of google-in-guest, but you don't have that context in the dial plan. |
00:55.52 | cusco | well the buddy calling is not a guest |
00:55.54 | p3nguin | And the usernames in jabber.conf are typically yourid@gmail.com/Talk |
00:56.12 | cusco | /Talk? |
00:56.50 | p3nguin | Did you bother reading the wiki that explains how to set it up so it works? |
00:58.42 | cusco | yes |
00:58.52 | cusco | well http://www.voip-info.org/wiki/view/Asterisk+Google+Talk |
00:59.03 | cusco | right /talk |
00:59.58 | cusco | tho it works dialing to buddies |
01:01.29 | cusco | so I did those two thingies |
01:01.35 | cusco | still not getting the call |
01:01.45 | cusco | i replicated the context and named it google-in-guest |
01:04.11 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
01:04.17 | cusco | wb |
01:04.21 | cusco | guess u didn't read |
01:04.50 | cusco | so I added the context (replicated from google-in and changed the name) and added /Talk |
01:04.55 | cusco | still not getting the call in |
01:04.57 | cusco | dunno why |
01:05.04 | cusco | it shows on jabber debug |
01:05.14 | *** part/#asterisk snuff-work (~snuffy@210.9.82.197) |
01:06.51 | cusco | it shows: to="merdosa@gmail.com/TalkD4B81223 |
01:07.00 | cusco | instead of /Talk |
01:07.02 | cusco | alone.. |
01:07.18 | cusco | I dunno what to look for |
01:07.35 | p3nguin | I followed the wiki and mine works. |
01:07.47 | cusco | :( |
01:09.18 | p3nguin | I didn't notice if you said you followed the wiki or not. |
01:09.44 | cusco | I read several howtos but I just checked everything from the wiki |
01:09.59 | cusco | I obviously missed te /Talk |
01:10.08 | cusco | I can make calls froma sterisk to gtalk |
01:10.20 | cusco | but I don't see incomming |
01:11.36 | p3nguin | Did you switch your calls to go to google chat only? |
01:11.45 | p3nguin | And don't login to chat in your email. |
01:12.36 | cusco | switch calls to go to google chat only? |
01:12.42 | cusco | didn't understand that question |
01:12.57 | cusco | I logged in at a time, but closed the browser (2 days ago) |
01:13.08 | cusco | I can login and click to terminate all other sessions... hold |
01:13.21 | p3nguin | When you login on voice.google.com, you go to the setup... |
01:13.30 | p3nguin | Go to the section where you set your phone number to forward calls. |
01:13.34 | cusco | but asterisk should be the only one logged in |
01:13.47 | p3nguin | Uncheck any forwarding numbers. Check only Google Chat. |
01:13.53 | cusco | ow... I don't think I did let me check |
01:14.12 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
01:14.43 | Naikrovek | it is amazing to me how much being an IRC op can get to someone's head... |
01:14.53 | Naikrovek | no one in here, though. at least not to my knowledge. |
01:15.29 | Naikrovek | one of the ops in #minecraft is *drunk* with 'power'. lol |
01:16.47 | cusco | I don't find any setup options on voice, besides the billing |
01:18.37 | p3nguin | Click on your GV phone number. Click on the Calls tab. |
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01:19.01 | cusco | yes I found that on google help, but no... wait I don't have a gv phone number |
01:19.06 | cusco | I need to click upgrade first? |
01:19.33 | p3nguin | I'm confused. How were you planning to get calls without a phone number? |
01:19.58 | cusco | err... I was trying only gtalk not google voice |
01:20.30 | cusco | so... gmail has chat box, and it allows talk (with a browser plugin) |
01:20.39 | cusco | I was trying to make that work with asterisk |
01:20.59 | cusco | and it works asterisk -> google talk |
01:21.18 | cusco | k let me click on upgrade |
01:21.21 | p3nguin | Ah, I thought you were asking about google voice. |
01:22.08 | p3nguin | I'd guess you'd have to make sure you don't have any other clients logged in, and then any calls to chat would arrive on your only logged-in client (asterisk). |
01:22.26 | cusco | ok Im guessing the same |
01:22.43 | cusco | jabber debug does read the incomming call |
01:23.44 | cusco | it says this is the only location |
01:23.46 | cusco | I logged out |
01:24.11 | cusco | ow |
01:24.14 | cusco | im getting something now |
01:24.21 | cusco | yay! |
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01:26.48 | p3nguin | Did you find out there was another client connected? |
01:27.10 | cusco | ok it works |
01:27.19 | cusco | no it said only me |
01:27.20 | cusco | ... |
01:27.22 | cusco | but well |
01:27.25 | cusco | that did something |
01:27.30 | cusco | logging in and clicking logout |
01:27.30 | p3nguin | What made it go? |
01:27.37 | cusco | that |
01:27.39 | cusco | lol |
01:27.44 | cusco | thanks for patience :D |
01:28.08 | cusco | now I still have to check this out when outside nat on both ends |
01:29.22 | cusco | and my objective is to be able to let people use gmail plugin as sort of a softphone |
01:30.33 | cusco | about google voice... |
01:30.45 | cusco | it costs money to have a number, right? |
01:30.52 | p3nguin | No. |
01:34.00 | cusco | hmm google is returning error in my request |
01:34.05 | cusco | will look at that another time |
01:34.57 | cusco | also... no jabber recieve chat unless i'm in a call right? |
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01:40.55 | cusco | http://i.imgur.com/76RkJ.png |
01:41.52 | p3nguin | I'm not sure, to be honest. |
01:42.29 | cusco | ok |
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02:59.36 | dandate2 | for some reason when a DID provider's forwarding is set to sip@ip my inbound routes don't recognize it, but if i remove the sip@ then it works? |
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03:34.58 | SeRi | p3nguin, you avail? |
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03:51.41 | p3nguin | seri: I'm here if you can get through the DoS. |
03:55.55 | SeRi | wtf |
03:55.58 | SeRi | seriously? |
03:56.15 | SeRi | p3nguin, whats going on? |
03:56.32 | SeRi | somebody DDoS you? |
03:56.48 | p3nguin | Disgruntled employee. |
03:57.01 | p3nguin | No, DoS. |
03:57.09 | p3nguin | DDoS is from multiple hosts. This attack is from a single host. |
03:57.18 | SeRi | bastard |
03:57.30 | p3nguin | It's just a high-bandwidth UDP flood. |
03:57.37 | SunTsu | p3nguin: then ask your ISP to blockhole that ip |
03:57.45 | SunTsu | blackhole even |
03:57.58 | p3nguin | I wish it were that easy. |
03:58.23 | SeRi | ^^ |
03:58.30 | p3nguin | seri: Yeah, once I can prove he's behind the attack, he'll probably be fired. |
03:58.35 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176001555.dsl.bell.ca) |
03:58.45 | SeRi | I called once comcast with a similar issue and they had no clue wtf whats going on lol |
03:58.53 | SeRi | they told me to email security... |
03:58.59 | p3nguin | Yep, they don't know anything. |
03:59.07 | p3nguin | Denial of what? :/ |
03:59.11 | SeRi | lmao |
03:59.24 | p3nguin | Or, "What kind of service?" |
03:59.41 | p3nguin | Denial of Service, sir. |
03:59.48 | p3nguin | "I'm sorry, we don't... have that." |
04:00.17 | SeRi | rofl |
04:00.24 | dijib | p3nguin, is the voip.ms customer portal getting ddos'ed or something or is it working for you? i cant get into order another iNum DID |
04:00.36 | SeRi | dijib, working for me |
04:00.49 | p3nguin | Nah, they're hammering one of my systems. |
04:00.49 | SeRi | I was just there getting a 5th number |
04:00.50 | dijib | srsly |
04:01.01 | dijib | who is? |
04:01.12 | p3nguin | I changed the IP address, so they are tracking it by my web site host name. |
04:01.31 | p3nguin | I'm not going to say his name until I have the proof in my hand that he is responsible for it. |
04:01.52 | dijib | yikes. then sue his ass |
04:02.11 | SeRi | dijib, not that simple... the law does not care unless you have a few mill in loses |
04:02.16 | p3nguin | He is pissed because I refuse to give him root access to a server that he used to have root access to. |
04:03.03 | dijib | i think the law... well here anyways would be interested in hearing the case even for a couple of hundred |
04:03.09 | SeRi | p3nguin, fuck him... look at him tomorrow and with an evil looked laugh at him and whisper (n00b) |
04:03.33 | p3nguin | The DoS is coming from utoronto.ca |
04:03.59 | p3nguin | I called the abuse phone number and left some messages, but got no response yet. I figure tomorrow I need to make a few more calls. |
04:04.00 | SeRi | dijib, seriously? cyber crime for a few hundred bucks? |
04:04.19 | SeRi | p3nguin, good luck. I know it can be painful |
04:04.20 | dijib | sure why not? at least small claims |
04:04.24 | dijib | utoronto.ca |
04:04.32 | dijib | is the person you think is there there? |
04:05.26 | SeRi | lol mhhhh ill stop there. read a bit about cyber crime. small claims can not touch it. |
04:05.48 | SeRi | any who p3nguin can you defer it with pf? |
04:05.52 | p3nguin | Once I can prove who is responsible, I will file criminal charges for endangering my family by DoSing my system where I have an internet phone system. |
04:06.04 | SeRi | well that you can do |
04:06.33 | p3nguin | I have better luck absorbing the packets rather than blocking them. |
04:06.34 | SeRi | rendering emergency services useless. Thta has a criminal intend |
04:06.39 | p3nguin | I accept them and do not reply. |
04:07.07 | p3nguin | By blocking them, the impact is greater. |
04:07.08 | SeRi | p3nguin, Cant be much. I mean most be coming from a consumer line given that you can still be online |
04:07.15 | SeRi | must* |
04:07.36 | p3nguin | No, it's from utoronto.ca |
04:07.36 | dijib | im 2hours from uofT |
04:07.52 | p3nguin | I'm sure they have a fairly large pipe. |
04:08.02 | SeRi | mhhhhhh |
04:08.40 | dijib | yeah one of the big pipes goes to front street which is about 3-6 blocks away |
04:08.58 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
04:09.26 | dijib | utoronto.ca is university of toronto or u of T |
04:09.33 | SeRi | is pondering |
04:09.46 | SeRi | p3nguin, shut the source down. |
04:09.47 | p3nguin | I knew he was upset, but I didn't know he was a packet kiddie. |
04:09.56 | SeRi | you can call the IT dep in the edu |
04:10.04 | dijib | cant you iptables fail2ban him? |
04:10.32 | SunTsu | dijib: as long as his uplink is filled up he can't do anything on his side |
04:10.47 | p3nguin | He's upset because his boss told me to change the root password and keep him off the server. |
04:10.47 | p3nguin | So I put it in lockdown. |
04:11.07 | SeRi | SunTsu, you beat me to it |
04:11.08 | SeRi | lol |
04:11.25 | dijib | then ifconfig eth0 down |
04:11.28 | SunTsu | U of Toronto = big pipe, p3nguin = considerably smaller pipe -> congestion -> bummer |
04:11.30 | dijib | no? |
04:12.01 | SeRi | dijib, look at what SunTsu is saying |
04:12.23 | SeRi | the only people that can put a stop at it is the ISP by defering the traffic or shutting the source down |
04:13.19 | SunTsu | p3nguin: I'd try to get in touch with your ISP's NOC/NMC/whatevertheycallit, ask them to nullroute it. 1st level support normally will put you through if you throw around terms they don't understand |
04:13.24 | SeRi | the damage is done. DoS is to take the target down by not allowing him to be online. some times it causes larger issue than that like crahsed systems etc... |
04:13.31 | p3nguin | I wouldn't be surprised if he's here watching what I'm saying about him. |
04:13.34 | dijib | network operations |
04:13.48 | p3nguin | He was on here every day until I didn't give him what he wanted, then he quit. |
04:14.50 | p3nguin | But when you're a moron who does not necessarily need to be touching computers, bosses tend to give other people your job. |
04:15.51 | SunTsu | p3nguin: looks like a wise choice to not have him being uid0 on your box |
04:16.01 | p3nguin | I use the interface he's hitting for connecting to IRC. If I shut it down, I won't be able to talk to you. |
04:16.46 | SeRi | that sucks... what a punk..... |
04:16.56 | SunTsu | p3nguin: and he'll prolly move on to another ip you own |
04:17.14 | p3nguin | Or as he put it, his box. |
04:17.17 | p3nguin | lamer |
04:17.29 | p3nguin | He has done two so far. |
04:17.41 | SeRi | p3nguin, well let me entreating you with some dial plan glory.... :P |
04:17.42 | SeRi | http://pastebin.com/Unu1Ur0r |
04:19.07 | SeRi | p3nguin, I hope you can catch the little fucker and burn him. |
04:19.09 | p3nguin | Two things to say about that. |
04:19.48 | SeRi | hahaha I knew it was all fucked up :P I thought I put it out there for you to laugh |
04:20.14 | SeRi | well I ddint know. I just knew you where going to say something :) |
04:20.19 | p3nguin | Line 3 does not make much sense to me. And Dial(Tech/peer/exten) |
04:22.19 | p3nguin | If you fix the pipe... line 3 says to me, if the DIALSTATIS is CHANUNAVAIL, go to a priority label the same as whatever number you just entered; if it is something else, go to the failover label, where you should change the syntax of the dial. |
04:22.24 | SeRi | well I am just trying to come up with a good way to fail over :) |
04:22.30 | SeRi | not doing a good job at it I guess :P |
04:22.56 | p3nguin | What will your failover be? |
04:23.07 | p3nguin | IAX2 on VoIP.ms? |
04:23.18 | SeRi | as a test yes |
04:23.23 | SeRi | just testing now |
04:23.35 | SeRi | eventually at some point in the extended future another provider :) |
04:23.44 | p3nguin | Dial(IAX2/voipms/1${EXTEN}) |
04:23.54 | p3nguin | Where voipms is a peer in iax.conf. |
04:24.25 | SeRi | yes I just didt it that way for now. I clear all the "cluster fuck" :) |
04:24.52 | p3nguin | If you don't have time to do it right the first time, when will you have time to redo it later? |
04:25.42 | SeRi | well I am just experimenting to see if it works. I have to learn some how ;) |
04:38.03 | drudge-gone | man |
04:38.09 | drudge-gone | .....effen china |
04:38.23 | drudge-gone | and thier effen stunts at 930pm |
04:38.42 | drudge-gone | i told this customer not to cheap out on a firewall |
04:38.59 | ChannelZ | the whole country should be disconnected from the net |
04:39.08 | drudge-gone | and afrinic |
04:39.11 | drudge-gone | agreed |
04:39.31 | drudge-gone | some customer got hacked |
04:39.43 | drudge-gone | i got called home from the bar, *sigh* |
04:40.42 | drudge-gone | they are lucky they dont pay for INTL or LD... locla only |
04:41.05 | drudge-gone | they are too small to afford some $15k+ phone bill for being hacked |
04:46.38 | dijib | nuke em |
04:50.21 | *** join/#asterisk SteveWilliams (~SteveWill@122.160.52.178) |
04:50.56 | SteveWilliams | Hi All! |
04:51.41 | SteveWilliams | Is there a way I can call an HTTP page from the dialplan in asterisk? |
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05:00.27 | [TK]D-Fender | SteveWilliams: System(curl http://....) |
05:03.51 | dijib | for what though? |
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06:16.04 | schmidts | good morning |
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08:02.45 | irroot | WIMPy ASTERISK_VERSION 10808 <- 1.8 but any SVN has 999999 this is confuzzeling |
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09:04.38 | *** join/#asterisk SteveWilliams (~SteveWill@122.160.52.178) |
09:05.03 | SteveWilliams | Hi All! Good Morning! |
09:07.34 | irroot | SteveWilliams top o the mornin |
09:10.58 | SteveWilliams | I would like to access the ${EXTEN} of a context from another context. Is that possible? Please help. |
09:11.47 | SteveWilliams | The other context is a macro and being called by the first context whose ${EXTEN} I would like to access. |
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09:18.41 | SteveWilliams | I would like to access the ${EXTEN} of a context from another context. Is that possible? Please help. |
09:18.44 | SteveWilliams | The other context is a macro and being called by the first context whose ${EXTEN} I would like to access. |
09:23.25 | SteveWilliams | ' |
09:27.03 | SteveWilliams | ' |
09:28.42 | Gugge | SET(IWANTTOUSETHIS=${EXTEN}) before you call the macro |
09:28.52 | Gugge | or something like that :) |
09:34.05 | irroot | indeed EXTEN is for the "line" in the dialplan its excuting at the time |
09:34.21 | irroot | you will need to set it or pass it as a ARG to the macro |
09:34.52 | irroot | Macro(mymacro, ${EXTEN}) |
09:35.14 | irroot | refer to it as ${ARG1} in the macro |
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09:36.16 | SteveWilliams | ' |
09:38.50 | SteveWilliams | <PROTECTED> |
09:39.08 | kaldemar | you don't need to pass it to the macro, just use MACRO_CONTEXT, MACRO_EXTEN, and MACRO_PRIORITY. they store the calling ones. |
09:39.23 | SteveWilliams | okay |
09:39.32 | kaldemar | what do you mean by access? |
09:40.20 | SteveWilliams | i am dialing a number from one context. when it connects, i pass it to another context which makes the callee listen to a voice message |
09:40.35 | kaldemar | and then? |
09:40.55 | SteveWilliams | i would like the macro retrieve the phone number and log it onto the database |
09:41.08 | SteveWilliams | the my sql database |
09:41.13 | kaldemar | why do you want the macro to do it? |
09:42.07 | SteveWilliams | i would like the macro to store the number in my mysql database along with the timestamp etc |
09:42.27 | kaldemar | SteveWilliams: i repeat, why do you want the macro to do it? why not do it in the extension? |
09:43.17 | kaldemar | if you store the things you want in the extension already, you can play a file with the A() option of Dial application. and you won't even need a macro. |
09:43.45 | kaldemar | but if you insist in using a macro, you have the original extension stored in MACRO_EXTENSION variable. |
09:43.54 | SteveWilliams | the macro gathers some data from the callee. i want that to be stored with the phone number |
09:44.05 | SteveWilliams | in my sql database |
09:45.04 | SteveWilliams | i have multiple files to play and various Read functions to go with them |
09:52.59 | SteveWilliams | Thanks, I used the MACRO_EXTEN. It works! Thank you kaldemar ! |
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10:09.30 | *** join/#asterisk Hyperbyte (jan@middelkoop.cc) |
10:11.09 | Hyperbyte | Hi! I have a dialplan for incoming calls, which dials all phones (peers) until someone picks up |
10:11.52 | Hyperbyte | I would like to make it so that, it doesn't try to call the phones that are already in a call. Right now when I'm in a call, and Asterisk receives another, it keeps trying to ring me. |
10:12.29 | Hyperbyte | I'm considering programming a feature into our softphones, which rejects calls when already in one, but I'd prefer to do this on the server side. |
10:14.09 | kaldemar | Hyperbyte: first Dial(Local/exten@contex&Local/exten2@context...) and then use GROUP functions in each extension to limit calls to 1. |
10:14.46 | kaldemar | sounds like you could a queue though. |
10:16.37 | Hyperbyte | kaldemar, if I limit calls to 1 like that, can they still make outgoing calls when already in a call? |
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10:32.20 | kaldemar | Hyperbyte: if you want to. |
10:33.21 | kaldemar | Hyperbyte: if you set and check a group only in the extension, it has no effect to calls from the phones. |
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10:45.30 | petern_ | hi, do any systems exist for speech recognition with asterisk, for data entry type situations where the input could feasably be anything, not waiting for a list a keywords... |
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11:22.12 | jacc0 | hoi all |
11:22.17 | jacc0 | almost weekend!!! |
11:22.24 | jacc0 | :) |
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11:29.19 | jsjc_ | hello I wonder I have this issue Unable to create channel of type 'DAHDI' (cause 0 - Unknown) |
11:29.23 | jsjc_ | because dahdi channel is already in use |
11:29.45 | jsjc_ | is there nyway I could in my dialplan create a condition if DIAL returns that error the dial trough this other channel? |
11:37.21 | ayrjola | check channel variable DIALSTATUS after dial |
11:41.20 | kaldemar | jsjc_: what is the other channel? |
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12:09.21 | jsjc_ | kaldemar: it is a sip channel. |
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12:12.40 | jsjc_ | ayrjola: what you mean by checking the variale after dial? could you give me an example? |
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12:13.12 | nunne | Does anyone know a good way to make asterisk issue a shell script/program etc. upon recieving a new voicemail (thinking of sms notification with the help of wget to a online sms service) |
12:13.58 | ayrjola | if dialstatus==CONGESTION, then DIAL(SIP/.....) |
12:14.28 | ayrjola | check in asterisk cli> core show application Dial |
12:14.44 | ayrjola | there you get information about DIALSTATUS values |
12:16.42 | ayrjola | if I remember correctly DIALSTATUS reports congestion if no DAHDI channels available |
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12:24.02 | nunne | ahh, just found externnotify.. no worries :) |
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13:00.13 | Vilius_Invade | hi Guys, is it possible to setup loopback SIP trunk? |
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13:02.06 | SteveWilliams | I have added a function to the func_odbc.conf file. Even after a cold reboot, asterisk says that the function does not exist. Please help |
13:03.37 | wdoekes2 | forget the prefix? |
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13:08.15 | SteveWilliams | I have added a function to the func_odbc.conf file. Even after a cold reboot, asterisk says that the function does not exist. Please help |
13:09.00 | SteveWilliams | ' |
13:11.21 | kaldemar | ~ask |
13:11.21 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
13:11.30 | kaldemar | show your dialplan and the function. |
13:12.53 | sehh | hey people |
13:12.55 | sehh | earlier today, I was asking for help with sending DTMF to the phone line, in order to enable/disable various features provided by my phone provider. I'm using ISDN lines with mISDN (chan_misdn). We found that its possible to do that with _MISDN_KEYPAD. Unfortunately, while this seems to work for Germany, it doesn't for me. The phone provider says that I've sent an invalid command. |
13:13.02 | sehh | any help would be appreciated please |
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14:01.25 | anonymouz666 | hello |
14:01.55 | anonymouz666 | 1.8.8.0-rc1 is a great version. |
14:02.16 | anonymouz666 | irroot: ping |
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14:02.37 | irroot | pong |
14:03.07 | irroot | indeed its pretty awesome had no issues reported by customers so far |
14:03.57 | anonymouz666 | irroot: I have seen one behaviour I think its a limitation about the feature wrapuptime used by app_queue, but when you have multiple agents and multiple queues, the wrapuptime is not "global" |
14:04.56 | anonymouz666 | the agents keep getting calls from queue A in the next sec, if they just hangup the call from Queue B. |
14:05.32 | irroot | ah that is something to be worked on indeed |
14:05.57 | puzzled | almost seems most questions are about queues these days |
14:06.23 | anonymouz666 | irroot: do you think that will be hard to do it, or do you know any patches that already does that ? |
14:07.22 | irroot | not know of any patches i put it on my work flow |
14:07.31 | jacc0 | I have some problems with timing in a brand new asterisk install (asterisk-1.8.7.0 + dahdi-2.5.0.1 on Debian) |
14:07.41 | *** join/#asterisk master_of_master (~master_of@p57B53526.dip.t-dialin.net) |
14:07.48 | jacc0 | Failed to open timing fd |
14:07.48 | jacc0 | Command 'timing test' failed. |
14:08.24 | jacc0 | I deselected timerfd in make menuselect |
14:09.22 | jacc0 | and selected dahdi |
14:09.24 | jacc0 | :S |
14:09.42 | jacc0 | I think it is the new dahdi failing |
14:10.34 | p3nguin | Works for me. |
14:10.41 | jacc0 | I've made the same configuation about 10 times in the past weeks; now I'm using the new dahdi and it fails |
14:13.01 | anonymouz666 | irroot: I have a setup under heavy load with exactly this problem, I would be glad to test when you finish, just let me know. |
14:13.34 | jacc0 | i use dahdi-dummy |
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14:14.23 | p3nguin | There's no dahdi dummy in dahdi 2.5.0.1. |
14:14.37 | irroot | module show like timing |
14:14.52 | anonymouz666 | but that would be a fix or a feature? |
14:14.56 | anonymouz666 | :P |
14:15.11 | irroot | make sure its not loaded maybe from previous build deselecting from menuselect does not stop old modules loading if installed |
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14:15.37 | irroot | dahdi_dummy is no longer standalone module dahdi core has the timing magic |
14:15.39 | jacc0 | i did : rm /usr/lib/asterisk/modules/res_timing_timerfd.so |
14:15.44 | irroot | so all you need is to load dahdi |
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14:16.25 | irroot | testing with faxes dahdi timing is best timerfd is not as good and pthread almost unusable |
14:17.07 | jacc0 | how do I 'load dahdi' |
14:17.17 | jacc0 | cli> load dahdi ? |
14:17.25 | p3nguin | modprobe dahdi |
14:17.51 | jacc0 | okay, so i need to : modprobe dahdi;modprobe dahdi-dummy |
14:18.01 | jacc0 | or can I leave out dahdi-dummy? |
14:18.12 | p3nguin | (0914.23) <p3nguin> There's no dahdi dummy in dahdi 2.5.0.1. |
14:18.40 | jacc0 | it doesn't give an error when I do: modprobe dahdi-dummy |
14:18.43 | p3nguin | If dahdi, timerfd, and pthread are all installed... If timerfd is not available (perhaps because it was unloaded), will dahdi be used automatically? |
14:19.09 | jacc0 | any other, no-exsisting name would fail |
14:19.24 | p3nguin | If you try to modprobe a module that does not exist, there should be some type of report saying it does not exist. That leads me to believe that you have old dahdi parts lying around. |
14:19.53 | p3nguin | modprobe -l dahdi\* |
14:20.25 | p3nguin | If you see dahdi/dahdi_dummy.ko, you have stale modules that should be uninstalled. |
14:21.12 | puzzled | old parts lying around is the reason why it makes sense to use a package management system like deb or rpm instead of installing src |
14:21.24 | p3nguin | Indeed. |
14:21.30 | jacc0 | dahdi/dahdi_dummy.ko is not there |
14:21.59 | kaldemar | from dahdi-base.c: MODULE_ALIAS("dahdi_dummy"); |
14:22.21 | p3nguin | Maybe modprobe doesn't report failures from missing modules or something. I really thought it did. |
14:22.30 | jacc0 | it does |
14:22.49 | jacc0 | but no error when doing modprobe dahdi-dummy |
14:22.58 | jacc0 | I'll try the older version |
14:23.28 | p3nguin | You'd be better off removing any old shit you have, installing the latest version, modprobe dahdi, and move on. |
14:24.05 | jacc0 | there is no old shit; it's a clean install |
14:24.24 | p3nguin | Good luck! |
14:24.30 | p3nguin | There's nothing else I can tell you. |
14:24.37 | jacc0 | okay, thnx |
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14:25.50 | kaldemar | maybe my alias comment didn't connect. dahdi_dummy is an alias for the core dahdi module. if you modprobe dahdi-dummy or dahdi_dummy, it just loads the core module. |
14:25.54 | devyll | question: what os is recommended for asterisk 1.8 ? centos ? debian? |
14:26.08 | kaldemar | devyll: one that you're comfortable with. |
14:26.13 | atheos | devyll the one you're most familiar with |
14:26.21 | devyll | ok centos 6? 5.5? |
14:26.48 | devyll | 5.7? |
14:26.58 | [TK]D-Fender | 8.6.7.5.3.0.9.? |
14:27.22 | devyll | :) got it |
14:27.24 | devyll | thanks |
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15:02.32 | twitchnln | morning |
15:03.12 | twitchnln | has anyone ever setup a grandstream gxw4104 with specific outbound rules as to which channel extension dials out on? |
15:03.12 | navaismo | morning |
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15:06.52 | navaismo | twitchnln use differents contexts |
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15:13.32 | FLeiXiuS | Is it possible for asterisk to use the same RTP ports if the client is connecting from the same IP address and the same SIP user? |
15:15.54 | ChannelZ | not that I'm aware of |
15:16.36 | [TK]D-Fender | no |
15:18.15 | FLeiXiuS | Im having an issue where two of my clients connect |
15:18.19 | FLeiXiuS | and hear the same voice call |
15:18.26 | FLeiXiuS | Even though they dialed 2 separate extensions |
15:18.47 | FLeiXiuS | 2 SIP users, dialing 2 separate conference rooms, listening to the same audio. |
15:19.18 | FLeiXiuS | Im lost on debugging ideas, figured they were using duplicate RTP ports |
15:19.50 | ChannelZ | not really possible |
15:19.59 | Naikrovek | you have the port range choked down too far, or you have a configuration problem, or you have some networking gear with real issues, or you have some networking gear that is trying to be "smart" and to "help" you. |
15:20.14 | ChannelZ | Are you SURE they're in different conferences? |
15:20.27 | Naikrovek | maybe someone bridged the conf's together. |
15:20.47 | FLeiXiuS | ChannelZ, yes, I can see meetme dropping them in 2 diff conferences. |
15:20.59 | FLeiXiuS | I minimized the config, very short only a few lines |
15:21.17 | p3nguin | What would cause some outbound calls through my ITSP to have normal ringing but other calls through the same ITSP, same iax peer entry, and same extension pattern to not have any ringing sound? |
15:21.20 | ChannelZ | Are both people behind the same firewall? |
15:21.21 | [TK]D-Fender | twitchnln, Go read the manual's section mentioning "Prefix To Specify Port" |
15:21.35 | FLeiXiuS | ChannelZ, yes, same machine. It's a web SIP client. |
15:22.13 | ChannelZ | two people are using the same computer? |
15:22.42 | FLeiXiuS | ChannelZ, They are connecting to the same web server, web server has a java applet which allows them to listen to a conference room. |
15:23.21 | FLeiXiuS | This applet is given instructions for which conf room to drop them in. |
15:23.36 | FLeiXiuS | IN asterisk, I can see the conf room being dialed and entered by the client. But the audio is exactly the same. |
15:24.18 | FLeiXiuS | BUT! When I reboot asterisk, the audio is different as you would expect. |
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15:26.02 | FLeiXiuS | If I dial from a soft sip client, the audio is different. It seems that connecting with the same server and the same sip user causes problems after like 10-20 calls |
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15:37.22 | jsarrel | dtmf signaling for analog lines is inband...correct? |
15:38.11 | irroot | jsarrel if by analog you refering to FXO POTS lines yes |
15:38.25 | jsarrel | yes, ty |
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15:59.24 | Greenlight | Howdy folks. Was wondering if someone could help with a question regarding jitterbuffers. The jitterbuffers configured using sip.conf and iax.conf, when should these be used, since most endpoints already have a jitterbuffer? |
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16:01.51 | Greenlight | Reason I'm asking is I've a couple of Asterisk boxes connected via IAX trunk and seem to be getting choppy and jittery audio |
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16:09.31 | navaismo | Greenlight maybe your bad audio its for bad network quality |
16:10.26 | Greenlight | Naa, it's only happening for calls into ConfBridge from the remote asterisk trunk. Direct calls are fine, and other calls into confbridge are fine |
16:10.58 | Greenlight | I've tried jitterbuffer on and off for the trunk and on and off for the confbridge |
16:11.42 | Greenlight | Was just wondering exactly what I should be setting - In my mind I only need the confbridge jitter buffer and nothing on the IAX trunk, but that doesn't seem to work ;/ |
16:12.37 | navaismo | enable trunking in the iax, use gsm codec or g729 to verify your network load isnt a issue |
16:13.03 | Greenlight | It's using trunked g729 already |
16:14.18 | Greenlight | Running at about 15% usage of the upstream bandwidth |
16:15.13 | Greenlight | Oh well - thanks anyways - Guess some more trial and error testing over the weekend :) |
16:16.32 | navaismo | np |
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16:17.20 | hardwire | meh |
16:19.58 | ChannelZ | FLeiXiuS: is this Java thing a server/daemon or do multiple instances just run as web clients fire it up? |
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16:26.06 | jblack | Hi. I seem to remember there's a problem with calling a macro from inside of macros. is that true? |
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16:58.51 | hardwire | sorta wants to make a cdr_cpickle |
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17:00.43 | FLeiXiuS | ChannelZ, nope, it runs locally on the users end in a JVM |
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17:35.24 | byronc | I'd like to create a dialplan extension from the console like this "dialplan add extension _X.,hint,${CUT(REALTIME(presence_hints,key,${CONTEXT}:${EXTEN},:,:),:,4)} into foo" but when I run the command I find an extension like this: '_X.' => hint: ${CUT |
17:35.33 | byronc | How would I escape that value correctly? |
17:36.10 | byronc | The real goal is to not have any of that substitution evaluate until a hint lookup on the context is done? |
17:36.54 | [TK]D-Fender | Bryanstein, You can't just put a function there. |
17:38.41 | [TK]D-Fender | Bryanstein, hints are fed on dialplan load and don't evaluate |
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17:43.09 | byronc | [TK]D-Fender: I have those hints working out of realtime: http://pastebin.com/yqxVLPcX |
17:43.39 | byronc | When a request is made for any hint on that context, it does a realtime lookup to find the right hint value |
17:44.06 | byronc | But I'd like to be able to add the generic hint from the console instead of having to run "dialplan reload" |
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17:50.37 | [TK]D-Fender | Bryanstein, Hrm |
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19:14.53 | Kastegir | hello, I am trying to get my TDM410P cards to work. I have fully functional FXO ports but no dial tone on the FXS ports. Can anyone help? |
19:21.18 | [TK]D-Fender | Kastegir, Did you plug the molex in? |
19:21.47 | Kastegir | yes, first thing I checked |
19:22.22 | [TK]D-Fender | pastebin your configs. |
19:22.25 | [TK]D-Fender | ? pb |
19:22.32 | [TK]D-Fender | ~pb |
19:22.32 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
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19:23.54 | Kastegir | I'm a little new at this, which config dop you want? |
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19:27.19 | [TK]D-Fender | all of the DAHDI one. |
19:27.21 | [TK]D-Fender | s |
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19:31.48 | Kastegir | just paste the link? |
19:33.58 | Kastegir | http://pastebin.com/hhBRYwNC |
19:34.53 | Kastegir | http://pastebin.com/kqVmzRGa |
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19:45.06 | navaismo | no fxo_ks configuration do you create the dadhi extension too? |
19:45.39 | [TK]D-Fender | Kastegir, #include chan_dahdi_additional.conf <--- where is this in a PB? |
19:45.59 | Kobaz | haha |
19:46.16 | Kobaz | when you change a route to a sip device in cisco call manager, it drops every call to that device |
19:46.23 | [TK]D-Fender | Kastegir, So far you have not defined an extension for those DAHDI channels |
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20:00.09 | Kobaz | [TK]D-Fender: isn't that great? |
20:00.23 | [TK]D-Fender | indeed |
20:00.27 | Kastegir | sorry |
20:00.29 | Kastegir | http://pastebin.com/GJ7rsZsu |
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20:03.21 | [TK]D-Fender | Kastegir, pastebin "dahdi show channels" from * CLI and "dahdi_cfg -vvvv" from OS CLI |
20:05.04 | Kastegir | http://pastebin.com/fL6E2fT3 |
20:06.05 | Kastegir | http://pastebin.com/nxj86f2D |
20:06.53 | [TK]D-Fender | Kastegir, "ls -la /etc/asterisk" |
20:09.06 | Kastegir | http://pastebin.com/qDsGnUu0 |
20:11.37 | [TK]D-Fender | root has perms in there. |
20:11.40 | [TK]D-Fender | not good. |
20:12.00 | Kastegir | I know, its a test box |
20:12.08 | [TK]D-Fender | chown -R asterisk:asterisk /etc/asterisk |
20:13.59 | Kastegir | done |
20:19.43 | [TK]D-Fender | restart * |
20:23.21 | Kastegir | seriously.... it was a permissions issue? |
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20:23.27 | Kastegir | that worked |
20:23.29 | [TK]D-Fender | Is it working? |
20:23.34 | [TK]D-Fender | That would do it |
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20:23.48 | Kastegir | I will song of your greatness by the fire.... thank you |
20:23.53 | Kastegir | sing even |
20:24.04 | [TK]D-Fender | * runs as "asterisk". fail to load the config that specifies that port due to privileges = fail |
20:24.15 | [TK]D-Fender | Kastegir, You're welcome |
20:24.51 | [TK]D-Fender | And on that note, it's checkout time. Later all... |
20:24.55 | Kastegir | didnt even occure to me that the file it made it couldnt read |
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21:21.05 | SuperNull | Hey Guys, anyone ever see asterisk have a megaton socket connects: asterisk 24873 root 504u sock 0,6 0t0 640986 can't identify protocol |
21:21.19 | SuperNull | errr i will pastebin |
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21:22.36 | dijib | p3nguin, still getting ddos'd |
21:22.37 | dijib | ? |
21:23.27 | p3nguin | DoS, not DDoS. And not right now. |
21:23.32 | SuperNull | might try fail2ban .. i have been waiting for the right situation of DOS. Could probably prevent a lot of failed login attempts |
21:23.39 | dijib | not distributed/ |
21:23.40 | dijib | ? |
21:23.51 | p3nguin | fail2ban is useless in this case. |
21:23.55 | dijib | thats what im saying SuperNull china or jp wouldnt have it lastnight |
21:23.59 | dijib | im like pppffft |
21:24.12 | dijib | bounce the interface and bam |
21:24.17 | dijib | more bacon |
21:24.21 | dijib | and bacon |
21:24.25 | SuperNull | swear to god.. our company got raped by russia telekom for like 40k in international .. |
21:24.49 | dijib | nuts |
21:24.53 | SuperNull | yeah. |
21:25.06 | SuperNull | we upgraded to 10gig and level3 said 'ehh okay it didnt happen' |
21:25.10 | dijib | how do you have a context security zone... im just home user nuub |
21:25.22 | dijib | ehhh canadiances? |
21:25.44 | p3nguin | Not distributed. |
21:25.54 | dijib | see i dont see the corporate itsp pricing schema |
21:25.57 | SuperNull | what are they doing p3nguin ? |
21:26.01 | p3nguin | Just a UDP flood from a single host. |
21:26.01 | dijib | one source. u of t |
21:26.19 | dijib | i think iptables and fail2ban would have worked |
21:26.21 | dijib | last night |
21:26.22 | SuperNull | is it saturating your uplinks ? |
21:26.24 | dijib | still flooding? |
21:26.34 | p3nguin | I called the CIO today and left a message. I then called the helpdesk and the minion was supposed to pass along the info to a manager. |
21:26.49 | p3nguin | It's saturating my downlink. |
21:26.54 | dijib | any inpact on the traffic now? |
21:27.07 | p3nguin | It's not happening right now. |
21:27.23 | SuperNull | multi-homed using bgp ? |
21:27.28 | p3nguin | And no, iptables and fail2ban would not have worked. Please don't recommend things you don't understand. |
21:27.34 | dijib | bgp? |
21:27.51 | dijib | im like shoot. lets go plinking |
21:27.52 | SuperNull | i do understand now .. obviously firewalling it would not prevent udp flooding. unless it was stopped pre-downstream router |
21:28.23 | dijib | well then im getting an elightenment bya considertion from jp |
21:28.30 | p3nguin | My remark was directed at dijib's suggestion that "iptables and fail2ban would have worked." |
21:28.31 | dijib | and no im not qualified |
21:28.53 | SuperNull | curious .. how much traffic ? |
21:29.01 | SuperNull | gigs or megs ? |
21:29.04 | p3nguin | gigs |
21:29.11 | SuperNull | yeah thats a whore of a problem to deal with. |
21:29.47 | dijib | i still dont understand why blocking that fqdn wouldnt work... but again im a nuuub |
21:30.08 | SuperNull | we had 12 gig coming at us one night.. because some kid pissed off someone on call of duty who had the ability to dos him.. 12 gigabit going to a 5megabit cable modem. |
21:30.42 | dijib | ability = bandwidth ? |
21:30.45 | p3nguin | dijib: You can block anything you want, but it still uses up your bandwidth to get it to you... where you block it. |
21:31.06 | dijib | ok then/// still need to say no... dont rape me |
21:31.07 | p3nguin | The only solution is to have it blocked up stream. |
21:31.15 | dijib | talk to the makers of IOS for that one |
21:31.35 | p3nguin | And iptables won't work because it's not a Linux box. |
21:32.02 | SuperNull | its not all bandwidth... with good transit providers they can mitigate things before they get to you.. but it may not be a quick thing it took level3 30 minutes from our call in |
21:32.13 | p3nguin | And fail2ban won't work because that's not what it does. fail2ban detects certain types of things in log files and responds by blocking traffic from the host. |
21:32.31 | p3nguin | But I already identified the host. |
21:32.40 | SuperNull | fail2ban utilizes iptables so its kind of a dillema ... i thought you had random peoples trying to hack sip accounts.. we get that all the time. |
21:32.51 | p3nguin | I can use fail2ban with pf if I want. |
21:32.54 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
21:33.04 | SuperNull | does it work with pf ? |
21:33.07 | p3nguin | Sure. |
21:33.10 | SeRi | p3nguin, you got good news for me? :) |
21:33.11 | SuperNull | i dont use *bsd so |
21:33.20 | p3nguin | seri: Maybe. |
21:33.27 | SeRi | ? |
21:34.04 | p3nguin | seri: The SIM arrived, but I'm not sure if it will do any good. It is my understanding that I'm supposed to use a SIM which was used to activate another iPhone. |
21:34.04 | SuperNull | my least favorite type of dos is email tho .. people get pissed at even a 5 minute delay in email delivery. |
21:34.20 | dijib | how would i build a security perimeter for outound calling in my dialplan using contexts? |
21:34.36 | dijib | get them using sip. |
21:35.13 | SeRi | p3nguin, Try it. report back. I never had to use an original sim to use ny iphone. |
21:35.19 | dijib | p3nguin, jailbrak that iphone h'esuze |
21:35.29 | SeRi | any* |
21:35.34 | dijib | jailbreak |
21:35.40 | SeRi | p3nguin, you still dealing with the DoS? |
21:36.16 | p3nguin | The DoS I get is just a bandwidth-consuming, CPU-hogging, log-filling type of attack. But I disabled logging, so that took care of log filling. I accept the traffic, so that took care of some of the CPU hogging. I accept the traffic but block replies, so that took care of part of the bandwidth consuming part. |
21:36.40 | p3nguin | Not yet today. He usually starts it at 6 PM Eastern time. |
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21:37.02 | SuperNull | why not block inbound so no reply is generated period. |
21:37.03 | p3nguin | I have an idea. :) |
21:37.27 | SeRi | p3nguin, let me also add that if that would be the case with your iphone than you are sol... that would be hard to find unless you buy a sim unlocker from china |
21:40.03 | SuperNull | anyone ever see anything like this from lsof output for asterisk: http://pastebin.com/qe04JnZn |
21:41.14 | SuperNull | we use mysql for realtime as well as some mysql lookups in dialplan im wondering if its getting stuck but.. oddly.. its all tcp not unix sockets. |
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21:44.14 | SuperNull | wonders if p3nguin did to much firewalling |
21:45.43 | *** join/#asterisk ChrisInSydney (~Chris@101.170.108.129) |
21:46.00 | SuperNull | Chris where are ya from ? |
21:46.56 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
21:47.57 | ChrisInSydney | Sydney |
21:48.16 | ChrisInSydney | Hey P3nguin: The bridge is was still up 10 minsago |
21:48.28 | ChrisInSydney | Eastern Beaches |
21:48.48 | ChrisInSydney | Somewhere between Bondi and Maroubra |
21:48.58 | SuperNull | rainin like a mofo here. |
21:48.59 | ChrisInSydney | Thats as close as I'll let ya |
21:49.13 | ChrisInSydney | yep. You in au too ? |
21:49.53 | SuperNull | nooo actually upstate New York.. |
21:49.55 | SeRi | p3nguin, you avail? |
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21:50.06 | ChrisInSydney | ahh |
21:50.10 | ChrisInSydney | rainin here too |
21:50.12 | ChrisInSydney | not too bad |
21:50.45 | ChrisInSydney | but enough to make the trip to swimming for the little one a little scary |
21:52.06 | ChrisInSydney | VUC is still up. 3 people |
21:52.19 | *** join/#asterisk devil_evoxxx (~d3v1l@host79-37-dynamic.4-87-r.retail.telecomitalia.it) |
21:52.24 | devil_evoxxx | hi all guy |
21:52.29 | devil_evoxxx | i've got a question... |
21:52.45 | ChrisInSydney | One is me at home, that leaves 2 other die hards |
21:53.01 | ChrisInSydney | devil_evoxxx: Ask away |
21:53.02 | devil_evoxxx | i serve to a small provider a sip interconnection for terminating sip traffic |
21:53.07 | devil_evoxxx | ahahah ok :) |
21:53.14 | ChrisInSydney | and... |
21:53.24 | devil_evoxxx | i bill the call after with a cron |
21:53.26 | devil_evoxxx | every 10 minute |
21:53.31 | SuperNull | uhg. |
21:53.33 | SuperNull | messy. |
21:53.38 | SuperNull | Radius baby. |
21:53.38 | devil_evoxxx | but, is not safe, because if the call remain up..i'm fucked |
21:53.46 | ChrisInSydney | yep |
21:53.51 | ChrisInSydney | take up music instead |
21:53.59 | ChrisInSydney | guitar. I'd recommend as a start |
21:54.08 | SuperNull | rockband for complete newbs. |
21:54.12 | SuperNull | plus you get drums too |
21:54.15 | ChrisInSydney | and you thunk I am joking :0) |
21:54.31 | devil_evoxxx | :( |
21:54.35 | devil_evoxxx | in my case |
21:54.50 | SuperNull | devil_evoxxx i can tell you the evils of text file based billing FIRST HAND. |
21:55.11 | devil_evoxxx | y |
21:55.13 | devil_evoxxx | yes |
21:55.16 | SuperNull | my company loses billing all the damn time because the previous guy implemented the most redic text based CDR system that merges everything.. monthly.. and then runs perl scripts against it |
21:55.18 | SuperNull | well |
21:55.28 | SuperNull | the way your doing it is even more painful |
21:55.39 | devil_evoxxx | the only wai for make it safe is using radius? |
21:55.43 | SuperNull | but possibly less likely to lose stuff than ours.. or lose long calls. |
21:55.57 | ChrisInSydney | I concur with SuperNull |
21:55.59 | SuperNull | ehhh radius is just real sexy for that kind of thing. its been around for ever.. and you can easily run searchs against it quickly |
21:56.34 | ChrisInSydney | I've not had much to do with billing, but the standard CDR stuff is not really your friend |
21:56.44 | devil_evoxxx | and i can send "disconnect" when is time.. |
21:56.48 | SuperNull | call starts - radius sends a call start. call stops radius sends a call stop .. and calculatues things for you.. then you just have to make reports instead of working on some kind of ghetto rigged text file situation |
21:57.03 | ChrisInSydney | Fine for stats, and aproximates, but billing is a nightmware |
21:57.13 | SuperNull | text based cdr is the devil for end points. |
21:57.21 | devil_evoxxx | there is some example for handlig this? |
21:57.26 | SuperNull | ehhh... |
21:57.27 | ChrisInSydney | Hey, some of the best music came out of the Ghetto |
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21:58.28 | SuperNull | devil_evoxxx all i can say is.. you will wish you could find me on irc again later to thank me if you are serious about not losing billed seconds. |
21:58.30 | devil_evoxxx | ehm but i've got another question |
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21:58.49 | ChrisInSydney | thats OK. Ask away |
21:59.02 | devil_evoxxx | SuperNull: is not the time for losing billed second..the problem that i've just observerd is |
21:59.29 | devil_evoxxx | on TRUNK_1 i receive the call from the other is |
21:59.31 | devil_evoxxx | isp |
21:59.48 | devil_evoxxx | that, i receive a call...and it'start |
22:00.07 | devil_evoxxx | (let 's suppose that |
22:00.32 | devil_evoxxx | the call duration is 240 second |
22:00.33 | SuperNull | supposes |
22:00.42 | devil_evoxxx | after 1 second when call 1 start, another call start |
22:01.05 | ChrisInSydney | ...and... |
22:01.19 | SuperNull | hey man .. bable fish cant translate that quick ;-) |
22:02.38 | p3nguin | seri: iTunes says I need to try again later or contact customer care. |
22:02.58 | p3nguin | In other words, it doesn't work. |
22:02.59 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
22:04.15 | ChrisInSydney | nope, they dont |
22:04.17 | ChrisInSydney | BRB |
22:04.28 | devil_evoxxx | i'm back |
22:04.55 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-qhwfreaaufkyntwd) |
22:05.19 | p3nguin | seri: http://support.apple.com/kb/ht3406 |
22:06.21 | devil_evoxxx | so, if i use a "prepayed" structure, how can i calculate the amount of second |
22:06.29 | devil_evoxxx | of two concurrent call? |
22:06.36 | devil_evoxxx | with one call , ok, no problem |
22:06.44 | devil_evoxxx | but with >2? |
22:10.35 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
22:10.56 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
22:12.55 | SuperNull | errr i cant speak about prepaid it is not a feature i know of for asterisk .. it may or may not exist as far as i know |
22:13.40 | devil_evoxxx | and, a question |
22:13.56 | devil_evoxxx | if your small client |
22:14.11 | devil_evoxxx | has been hacked , and make a lot of calls |
22:14.15 | devil_evoxxx | how can you prevent this? |
22:15.29 | *** join/#asterisk Fritz09 (~Adium@pop1-1489.catv.wtnet.de) |
22:16.37 | SuperNull | set a call max ? or calls per second limit some how.. when we got hacked it was usually long duration international coming from russia or china |
22:16.49 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
22:16.55 | SuperNull | 20 hour call .. like .. seriously.. who does that. |
22:17.02 | devil_evoxxx | in my case the ip was from palestina |
22:17.07 | devil_evoxxx | and calla special service in Estonia |
22:17.15 | SeRi | p3nguin, I think you have no choice but to jailbrake your phone and unlock it. |
22:17.15 | devil_evoxxx | 1000 Euro burned.. |
22:17.31 | ChrisInSydney | Ouch :-( |
22:17.39 | devil_evoxxx | because |
22:17.42 | devil_evoxxx | this fucking client |
22:17.47 | SeRi | I never came across any issues with unlocked iphones. |
22:17.47 | devil_evoxxx | don't take care of his hata |
22:17.49 | devil_evoxxx | ata |
22:17.55 | SeRi | none unlocked* |
22:18.10 | devil_evoxxx | and..all know that with an exploit you can extract password |
22:18.13 | devil_evoxxx | and user.. |
22:18.50 | devil_evoxxx | it's saying to me that all ata was on default port..with default password |
22:18.52 | devil_evoxxx | FUCK.. |
22:18.56 | devil_evoxxx | i have no word. |
22:19.10 | SuperNull | bill em. nuff said. not your problem. |
22:19.57 | SuperNull | not like your provider will give it to ya free. |
22:20.10 | devil_evoxxx | eh, it's right |
22:20.13 | SuperNull | unless you upgrade from 1gigabit ethernet to 10gigabit .. then maybe.. |
22:20.38 | devil_evoxxx | ..oO we'have recently upgraded to 1 gbit STM-16 |
22:20.53 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
22:21.03 | SuperNull | where is this and .. they dont have native ethernet options yet/ |
22:21.10 | p3nguin | seri: It can't be jailbroken, it can't be activated without the SIM it's looking for. |
22:21.25 | devil_evoxxx | SuperNull: ethernet over sdh too.. |
22:21.25 | p3nguin | seri: Once I can activate it, I can probably jailbreak it. |
22:22.39 | SuperNull | devil_evoxxx i think your best bet is finding a solution or really planning it out.. it sounds like a lot of work possibly with AGI .. and im guessing because i dont know AGI |
22:22.41 | SuperNull | haha |
22:23.22 | devil_evoxxx | i know agi , i've already use |
22:23.30 | SeRi | p3nguin, http://www.iphonehacks.com/2010/07/how-to-activate-your-iphone-without-official-iphone-carrier-sim.html |
22:23.39 | SeRi | Read that ^^ |
22:23.48 | SuperNull | so if its prepaid why even allow multi-calling at the same time ? |
22:24.18 | SuperNull | why not just say 'sorry a call with that pin is already active.. go eat a cake' |
22:24.38 | SuperNull | using radius this would be fairly easy.. |
22:25.12 | SuperNull | then you just have to worry about max duration of a single call and can probably somehow make that drop after a certain max time. |
22:25.20 | devil_evoxxx | SuperNull: because is not only a phone |
22:25.22 | devil_evoxxx | is a trunk |
22:25.25 | devil_evoxxx | type=peer |
22:25.32 | SuperNull | prepaid trunks ? |
22:25.44 | SuperNull | ehhh we know of no such thing in the USA. |
22:25.51 | devil_evoxxx | y.. |
22:26.10 | SuperNull | in Soviet Russia VOIP trunk pay for you. |
22:26.13 | SuperNull | ehh |
22:26.17 | devil_evoxxx | if you give a service to a small providere for sip-trunking |
22:26.21 | SuperNull | we bill on usage only and cross our fingers they pay. |
22:26.52 | devil_evoxxx | how you do= |
22:26.52 | devil_evoxxx | how you do? |
22:27.10 | SuperNull | we dont do that type of call service.. we are primarly a VOIP provider for residential docsis modem customers. |
22:27.21 | SuperNull | about 8000-9000 to be exact |
22:28.22 | dijib | my brother bought an iphone 4S today. ive suggested he runs an asterisk implementation on it. |
22:28.30 | SuperNull | hahaha |
22:28.37 | SuperNull | do they have such a thing dij? |
22:28.47 | dijib | you could build it yes. |
22:28.55 | dijib | positive |
22:28.57 | SuperNull | if you build it.. they will call |
22:29.02 | dijib | its got a 1ghz dual core cpu |
22:29.15 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
22:29.26 | dijib | thats what im saying... callcenter in a phone. call all your service departments off a queue |
22:29.36 | SuperNull | so .. thats an interesting concept.. |
22:29.45 | SuperNull | phone switch on the phone. |
22:29.51 | dijib | word up. mobile terminal scripsts for adding peers |
22:30.12 | SuperNull | any ideas if the wifi chip in that thing does WDS ? |
22:30.13 | dijib | he told me he mainly uses it for email sms and web. i said... get an account pat. |
22:30.24 | dijib | lol |
22:30.29 | SuperNull | how cool would that be.. |
22:30.39 | SuperNull | mesh of iphone .. phone switches.. |
22:30.43 | dijib | that just is passive mode on the wifi no? |
22:30.59 | SuperNull | WDS is like.. an ap that links to another ap .. and .. can still accept users.. |
22:31.03 | dijib | wds is linksys or cisco tech i think |
22:31.30 | SuperNull | ehh its an open standard i believe .. if its not.. its documented enough to be supported by many |
22:31.39 | dijib | no he would be using the telco's bandwidth... but im sure if there is a less costly path the routing protocal would handle the switchj |
22:31.52 | SuperNull | i used to use it with HostAP years ago.. when commercial 802.11 APs were like $1000 |
22:32.04 | dijib | oh |
22:32.10 | SuperNull | you could dij.. |
22:32.17 | dijib | diji |
22:32.18 | SuperNull | anyone in range could hop you to the next person in a city.. it would be intense.. |
22:32.24 | SuperNull | diji ... |
22:32.24 | dijib | as in digital equipment corporation |
22:33.01 | SuperNull | the delimma would be routing tho.. i have used AODV routing before in a beta format.. and its slow as poo. |
22:33.20 | dijib | so what do you think, theory on the iphone asterisk server good? also give it dydns |
22:33.53 | SuperNull | iphone + thin asterisk + wds + dynamic on demand routing |
22:33.59 | p3nguin | waves at chrisinsydney |
22:34.16 | SuperNull | honestly.. i thought of this before.. for android.. but i dont want to brick my Evo 4G :-/ |
22:34.43 | dijib | at least the iphone is restorable easily |
22:34.58 | dijib | i would love to alpha on his old iphone 4 |
22:35.01 | SuperNull | i would be curious to see what kind of call quality you could get on 3G .. lol |
22:35.36 | dijib | seeing that your client would be connecting to 127.0.0.1 i would hope good in the 3g band |
22:35.50 | dijib | thats what telco uses anyways.. obviously |
22:36.14 | devil_evoxxx | guys, i left...good night |
22:36.19 | SuperNull | yeah but they can setup QOS service flows of some type im sure, so they have covered their butt |
22:36.42 | SuperNull | im leaving for the weekend as well Gents.. i will likely be back when this asterisk box shits from to many open sockets. |
22:36.59 | dijib | no way with the service being split with wireless internet packages |
22:37.20 | dijib | have a good one SuperNull |
22:44.53 | ChrisInSydney | The VUC bridge is still up for you die hards |
22:46.47 | ChrisInSydney | SIP:200901@login.zipdx.com (g722, g711) |
22:57.14 | p3nguin | I got pulled into making supper, so I'll be away from the conf for a little while. |
22:58.56 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-nbfnybjamzytdtpx) |
22:59.18 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176001555.dsl.bell.ca) |
23:01.40 | p3nguin | seri: Did you read that page or watch the video? |
23:02.08 | ChrisInSydney | P3nguin. Mans gotta eat |
23:02.48 | p3nguin | I intend to do just that. |
23:02.59 | ChrisInSydney | I'm still hacking the data. Cant get crimson editor to do reg expr :-( |
23:03.06 | ChrisInSydney | oh well, it is free |
23:03.52 | ChrisInSydney | So whats for dinner ?? |
23:03.54 | dijib | so asterisk security perimeteres? |
23:04.14 | dijib | chicken leggs and baked potatoes... shaddup beach |
23:04.23 | ChrisInSydney | dijib perimeters or parameters ? |
23:04.30 | p3nguin | Looks like spaghetti, meat balls, and garlic bread. |
23:04.44 | ChrisInSydney | and a glass of something red |
23:05.16 | ChrisInSydney | dijib: SIP:200901@login.zipdx.com (g722, g711) |
23:05.23 | ChrisInSydney | VUC bridge is still up. |
23:05.32 | ChrisInSydney | drop in and have a chat |
23:06.09 | p3nguin | ? |
23:06.15 | p3nguin | What do you mean what happened? |
23:06.40 | ChrisInSydney | felix is out of my sight at the moment. I heard a crash and a whine |
23:06.53 | ChrisInSydney | there is still noise so its not too bad |
23:07.06 | ChrisInSydney | no need to call DOCS (CPS) |
23:07.29 | ChrisInSydney | He's found his "In The Night Garden" book. hes happy now |
23:08.34 | dijib | i just want nobody but someone that authenticated be able to access voipms-outbound context |
23:08.42 | dijib | and ChrisInSydney whats this ? mailto:SIP:200901@login.zipdx.com |
23:09.02 | dijib | how did mailto: get there |
23:09.23 | ChrisInSydney | dont know |
23:09.28 | ChrisInSydney | dial in on SIP |
23:09.42 | dijib | a zip? |
23:09.47 | dijib | wtf is this p3nguin ? |
23:09.47 | ChrisInSydney | SIP:200901@login.zipdx.com |
23:10.05 | dijib | SIP/200901@login.zipdx.com |
23:10.28 | dijib | ? |
23:10.30 | SeRi | p3nguin, watch the video |
23:10.33 | ChrisInSydney | Dial(SIP/200901@login.zipdx.com) |
23:11.22 | p3nguin | seri: Did you? |
23:12.17 | SeRi | o no did not watch the video |
23:12.25 | SeRi | why? |
23:12.34 | SeRi | whats going on? |
23:12.41 | p3nguin | There's a big red note at the bottom that says the method is worthless as of a year ago and cannot be used now. |
23:13.00 | SeRi | o damn |
23:13.09 | SeRi | That sucks. |
23:14.12 | SeRi | p3nguin, http://www.google.com/search?aq=0&oq=jailbraking+iphone+without&sourceid=chrome&ie=UTF-8&q=jailbreaking+iphone+without+sim+card |
23:14.32 | p3nguin | chrisinsydney: Who was that person who came on? |
23:15.44 | ChrisInSydney | I think that was Dave Frankl |
23:15.50 | ChrisInSydney | runs ZIPDX |
23:15.53 | p3nguin | Oh. |
23:18.50 | dijib | ChrisInSydney, dialing that doesnt work |
23:18.59 | ChrisInSydney | what happens |
23:19.06 | dijib | exten => 6969,1,NoOp(); |
23:19.07 | dijib | same => n,Dial(SIP/200901@login.zipdx.com); |
23:19.14 | dijib | [Oct 14 19:07:40] == Using SIP RTP CoS mark 5 |
23:19.22 | dijib | busy beep |
23:19.22 | ChrisInSydney | check the codecs |
23:19.26 | *** join/#asterisk jblack (~jblack@pool-71-173-1-251.sctnpa.east.verizon.net) |
23:19.28 | dijib | im using ulaw |
23:19.31 | dijib | 711 |
23:19.33 | ChrisInSydney | should be OK |
23:19.41 | ChrisInSydney | whats in the cli |
23:20.02 | dijib | that last is in the cli |
23:20.07 | ChrisInSydney | I'll check my configs |
23:20.36 | dijib | p3nguin, how am i screwing up with that... and mind im drinking. |
23:21.10 | ChrisInSydney | exten => 882,1,Dial(SIP/200901@login.zipdx.com) |
23:21.33 | p3nguin | :) I use 882 as well. |
23:22.00 | p3nguin | Maybe it won't let in new calls? |
23:22.45 | ChrisInSydney | I jts checked. It does |
23:22.50 | ChrisInSydney | hjust |
23:22.54 | ChrisInSydney | just |
23:22.54 | p3nguin | Nope, it let me call in again. It even gave me a warning that said my ID was in use. |
23:23.08 | ChrisInSydney | I have a differnt ID |
23:23.14 | p3nguin | I only have one. |
23:23.17 | ChrisInSydney | actually, I'm not registered |
23:23.23 | p3nguin | I dial in with my PIN. |
23:24.24 | ChrisInSydney | oops. My son has just plugged in the barcode reader |
23:24.43 | ChrisInSydney | 606449067835 |
23:24.48 | *** join/#asterisk dandate2 (~dan@124.6.157.210) |
23:24.51 | ChrisInSydney | he he he |
23:25.06 | ChrisInSydney | 606449067835 |
23:25.14 | p3nguin | haha |
23:25.28 | ChrisInSydney | CHAE44025B |
23:25.37 | p3nguin | The first time I plugged in mine, I had IRC in focus and it did the same thing. |
23:25.37 | ChrisInSydney | I'm going to get him a storemans job |
23:25.57 | p3nguin | I think I printed my wife's driver license number in IRC. |
23:26.02 | ChrisInSydney | oops |
23:26.07 | p3nguin | :/ |
23:26.13 | p3nguin | I didn't know it was going to do that. |
23:26.24 | ChrisInSydney | thewse are just prouct serials |
23:26.25 | p3nguin | I plugged it in and started looking for things to scan. |
23:26.28 | dandate2 | so i got a problem, i got this DID that my inbound routes won't recognize, in core show channels it just says SIP/64.136.174.30-00 |
23:26.41 | p3nguin | That's normal. |
23:26.47 | p3nguin | So what's the actual problem? |
23:26.49 | dandate2 | how can i get inbound routes to recognize it? |
23:27.10 | dandate2 | it just sends it to the default any DID route |
23:27.28 | p3nguin | Are you sending SIP registrations to that ITSP? |
23:27.38 | p3nguin | And we don't know what "inbound route" means. |
23:27.44 | ChrisInSydney | dandate2: exten => 882,1,Dial(SIP/200901@login.zipdx.com) |
23:27.50 | dandate2 | oh its a freepbx term |
23:27.54 | dandate2 | i guess i should go there! |
23:27.55 | p3nguin | ~freepbx |
23:27.55 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
23:28.11 | p3nguin | I can help you fix the peer in asterisk. |
23:28.11 | ChrisInSydney | The VUC bridge is still up |
23:28.14 | ChrisInSydney | join us |
23:28.46 | p3nguin | I figure you just need to send the correct extension in your register statement and create said extension in the relevant context to accept calls. |
23:29.19 | p3nguin | Usually the extension is the same as the DID number, i.e. your phone number. |
23:30.22 | dandate2 | right, i experienced this problem with another ITSP. if the DID was forwarded as #@ip it worked, but if it was sip:#@ip it wouldn't |
23:30.26 | dijib | i think i need to register asd a peer in sip.conf no? |
23:30.46 | ChrisInSydney | correct. Or have a friend and user |
23:31.43 | p3nguin | You don't need a peer entry in sip.conf to call a SIP URI. |
23:32.26 | p3nguin | If you wanted to do Dial(SIP/zipdx/200901), then you'd need a peer entry for zipdx. |
23:32.30 | ChrisInSydney | You should be able to call a SIP URI that support anonamous connects |
23:32.46 | ChrisInSydney | so long as it is as a SIP URI |
23:32.53 | p3nguin | To dial the URI, just dial it like chrisinsydney said. |
23:32.59 | ChrisInSydney | SIP/name@host.whatever |
23:33.06 | ChrisInSydney | exten => 882,1,Dial(SIP/200901@login.zipdx.com) |
23:33.29 | ChrisInSydney | if you have freepbx, you can stick it into a custom extension |
23:33.43 | ChrisInSydney | or hack the dial plan. Just dont apply any more extensiosn |
23:35.33 | ChrisInSydney | dijib: Whats happening ? Still no luck ?? |
23:35.36 | p3nguin | If it still doesn't work, core set verbose 3, make a call to it, copy/paste the stuff in the pastebin. |
23:35.53 | p3nguin | If that doesn't show where the problem is, then we'll go on to the sip debug. |
23:36.58 | p3nguin | I hear someone talking from the other room, but I don't know what it is. I hope you weren't talking to me; I'm near the kitchen. |
23:37.06 | ChrisInSydney | nope |
23:37.17 | ChrisInSydney | [probably just to myself |
23:37.39 | p3nguin | or the child |
23:37.43 | ChrisInSydney | maybe it was the voices in my head. If they tell me to act normal, things seem to go OK |
23:39.49 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
23:40.34 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
23:40.39 | ChrisInSydney | drmessano: Join us. |
23:40.54 | ChrisInSydney | exten => 882,1,Dial(SIP/200901@login.zipdx.com) |
23:41.00 | ChrisInSydney | VUC bridge is still up |
23:41.11 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
23:41.13 | p3nguin | I'm still wondering about the "recording" thing that was mentioned. |
23:41.23 | p3nguin | I don't understand why it would be recording. |
23:41.37 | dijib | what recording thing? |
23:41.58 | dijib | still no luck ChrisInSydney |
23:42.03 | ChrisInSydney | paste the CLI |
23:42.15 | p3nguin | Good luck getting that from him. |
23:42.22 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-ohgcodxihklyoaui) |
23:42.27 | ChrisInSydney | dijib: the VUC is recorded |
23:42.29 | dijib | such a pesemist |
23:42.31 | p3nguin | I've decided to ask once and if I don't get it, fuck it. |
23:42.35 | ChrisInSydney | They pick the best bits out for the podcast |
23:42.38 | dijib | [Oct 14 19:12:54] -- Registered SIP '300' at 74.198.9.174:49941 |
23:42.38 | dijib | [Oct 14 19:16:54] == Using SIP RTP CoS mark 5 |
23:42.38 | dijib | [Oct 14 19:31:29] == Using SIP RTP CoS mark 5 |
23:42.39 | dijib | flood |
23:42.40 | p3nguin | I don't care enough to keep asking for it. |
23:43.03 | ChrisInSydney | strange |
23:43.19 | ChrisInSydney | what is the order of codecs in your SIP.conf |
23:43.46 | p3nguin | I'd rather see something relevant -- what I asked for. |
23:44.49 | dijib | what did you ask for ? |
23:45.28 | p3nguin | (1835.29) <p3nguin> If it still doesn't work, core set verbose 3, make a call to it, copy/paste the stuff in the pastebin. |
23:45.54 | *** join/#asterisk PhoenixMage (~Phoenix@CPE-120-146-192-94.static.vic.bigpond.net.au) |
23:46.55 | p3nguin | You know, standard procedure to get help. |
23:47.05 | PhoenixMage | Hi guys, I am having probs with a Cisco 7975 with SIP firmware. It can receive calls fine but when I try to make a call it attempts to dial but fails on the first digit as if that is the whole number |
23:47.18 | PhoenixMage | Same deal on dialing back a missed call |
23:47.39 | PhoenixMage | Any ideas what I am forgetting |
23:48.15 | p3nguin | Do not lift the handset. Dial the number, then press the Dial key. |
23:48.18 | p3nguin | Does that work? |
23:48.33 | PhoenixMage | nope same issue |
23:48.51 | ChrisInSydney | jimmy*CLI> core set verbose 10 |
23:48.51 | ChrisInSydney | Verbosity was 28 and is now 10 |
23:48.51 | ChrisInSydney | <PROTECTED> |
23:48.51 | ChrisInSydney | <PROTECTED> |
23:48.51 | ChrisInSydney | [Oct 15 10:47:52] NOTICE[1887]: chan_sip.c:12724 handle_response_peerpoke: Peer 'DBC-extn201' is now Reachable. (76ms / 2000ms) |
23:48.52 | ChrisInSydney | <PROTECTED> |
23:48.53 | p3nguin | core set verbose 3 |
23:48.54 | ChrisInSydney | <PROTECTED> |
23:48.58 | ChrisInSydney | <PROTECTED> |
23:48.58 | p3nguin | make a call. |
23:48.59 | ChrisInSydney | <PROTECTED> |
23:49.01 | ChrisInSydney | <PROTECTED> |
23:49.03 | ChrisInSydney | [Oct 15 10:48:00] NOTICE[1887]: chan_sip.c:15820 handle_request: Unknown SIP command 'UPDATE' from '76.74.151.123' |
23:49.05 | p3nguin | Why the flood? |
23:49.08 | p3nguin | ~pb |
23:49.08 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
23:49.14 | p3nguin | pastebin ^^^ |
23:49.58 | ChrisInSydney | I recon its codecs |
23:50.28 | p3nguin | Until he starts the troubleshooting procedure that we all use here, I don't care to guess what the problem is. |
23:50.31 | ChrisInSydney | <PhoenixMage> Jump onto the VUC bridge. Its still up |
23:51.00 | ChrisInSydney | <PhoenixMage>: exten => 882,1,Dial(SIP/200901@login.zipdx.com) |
23:51.22 | PhoenixMage | [Oct 15 10:50:35] NOTICE[17332]: chan_sip.c:21358 handle_request_invite: Call from 'button1contact' to extension '3' rejected because extension not found in context 'LocalSets'. |
23:51.33 | PhoenixMage | the number i am dialing is 3610 |
23:51.56 | ChrisInSydney | PhoenixMage: dialplan.xml ?? |
23:52.20 | ChrisInSydney | brb |
23:52.26 | PhoenixMage | ChrisInSydney: using the same context and dialplan for my iphone and ipad without issue |
23:52.28 | p3nguin | If you're not getting a dial tone first, but dialing the number and then pressing the Dial key, the dialplan file should not be consulted. |
23:52.44 | p3nguin | Your iPhone and iPad do not use the Cisco phone dialplan xml file. |
23:53.07 | PhoenixMage | I dont have a dialplan.xml |
23:53.13 | p3nguin | Maybe you should get one. |
23:53.20 | PhoenixMage | didnt know I needed one |
23:53.30 | PhoenixMage | everything talks about the SEP.xml files |
23:53.35 | PhoenixMage | any doco? |
23:53.39 | p3nguin | SEP is something else. |
23:54.24 | ChrisInSydney | I'll find some. I'll be back. |
23:54.36 | ChrisInSydney | Join us on the bridge if you can |
23:54.39 | p3nguin | The dialplan file tells the phone when to "send" the call, based on the number entered. |
23:54.49 | ChrisInSydney | I'm the last ine left and its getting lonely :-/ |
23:55.01 | p3nguin | If you don't have one, it should just take a really long time to send the call. |
23:55.11 | ChrisInSydney | need to attend to family |
23:55.13 | ChrisInSydney | 5 mins |
23:55.56 | *** join/#asterisk chalcedony (~llhull@unaffiliated/chalcedony) |
23:56.04 | PhoenixMage | I'll bbs I have to get my haircut |
23:56.11 | PhoenixMage | thanks for the help though |
23:56.27 | ChrisInSydney | <PhoenixMage>: There is some stuff on 7970s in the voip-info website. Explains dialplan.xml |
23:56.47 | p3nguin | Not having one should not cause the described problem, though. |
23:56.56 | ChrisInSydney | hair cut ?? |
23:57.02 | ChrisInSydney | h ehe he |
23:57.03 | p3nguin | But it will be nice to have it later, when things are working correctly. |
23:57.08 | PhoenixMage | yep I am overdu |
23:57.09 | PhoenixMage | e |
23:57.14 | ChrisInSydney | helps you see what you're dialling |
23:57.48 | ChrisInSydney | nah. Let it grow. Goi hang oiut on wall street |