| 00:05.37 | dexpdx | PoE ports? |
| 00:05.41 | dexpdx | power over ethernet? |
| 00:06.20 | dexpdx | I use switches with PoE and x86 hardware, but I don't use PoE |
| 00:07.03 | RMod | never seen a nic that provides PoE |
| 00:07.52 | MikeJ | ManxPower: jamaca man |
| 00:12.06 | *** join/#asterisk djeff (n=geoff@modemcable084.197-80-70.mc.videotron.ca) |
| 00:12.19 | djeff | hi |
| 00:12.53 | MikeJ | ManxPower: more interesting is Mexico isn't part of NANPA anymore |
| 00:13.51 | drmessano | Not since the mexican govt ordered the breakup of Taco Bell |
| 00:14.00 | *** join/#asterisk UnixDawg_ (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 00:15.40 | LiNeTuX_Home | How many folks in here have replaced a PBX and introduced the new Asterisk-based system to their staff? |
| 00:16.12 | MikeJ | US, Canada, Anguilla, Antigua, Bahamas, Bermuda, British VI, Cayman Islands, Dominica, Dominican Republic, Grenada, Jamaica, Montserrat, Saint Kitts and Nevis, Sait Lucia, Saint Vincent, Trinidad and Tobago, Turks and Caicos... plus Guam, Northern Mariana, Puerto Rico, US VI, American Samoa |
| 00:16.16 | MikeJ | thats quite a list |
| 00:16.18 | UnixDawg_ | asterisk is far from new and man have |
| 00:17.47 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 00:18.33 | MikeJ | s/man/many/ |
| 00:19.20 | *** join/#asterisk Steve_J-obs (n=Chris123@wsip-24-234-220-113.lv.lv.cox.net) |
| 00:19.57 | UnixDawg_ | thanks |
| 00:20.08 | UnixDawg_ | going to bed been a long day |
| 00:22.18 | Steve_J-obs | hello guys, anybody please give an idea why my sip client cant register with the new router (linksys wrt610n) that I bougth? |
| 00:22.34 | hardwire | what sip proxies do you guys use for CPE? |
| 00:29.33 | Steve_J-obs | my freaking router is not letting my sip client registter, anyboy plese any ideas? |
| 00:34.22 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 00:38.40 | MikeJ | Steve_J-obs: shouldn't you be using an airport Extreme? That should fix it |
| 00:38.47 | drmessano | lol |
| 00:39.02 | drmessano | Why dont you go ask your IPOD |
| 00:39.11 | drmessano | sorry |
| 00:39.15 | v4mp | evening drmessano |
| 00:39.59 | MikeJ | Steve_J-obs: seriously.. your trying to register "with" your router? like it has a sip registrar in it? |
| 00:40.02 | drmessano | dude.. I so thought about you at work today |
| 00:40.28 | drmessano | I was IM'ing from my Blackberry, sitting on the crapper.. and it was like Deja Yu |
| 00:40.40 | v4mp | drmessano, u thought about me ? |
| 00:41.16 | MikeJ | from the toilet? |
| 00:41.34 | Steve_J-obs | mikeJ: I am trying to register to an asterisk server... but this new router has something on it, that's not letting it |
| 00:41.50 | Steve_J-obs | ...and it is not the firewall |
| 00:41.58 | drmessano | Try #linksys |
| 00:41.59 | MikeJ | so asterisk inside the router, sip client outside? |
| 00:42.02 | Steve_J-obs | linksys wrt610n |
| 00:42.15 | drmessano | Where is Asterisk located? |
| 00:42.18 | Steve_J-obs | no no... asterisk outside |
| 00:42.40 | MikeJ | is the register ever getting to the asterisk? |
| 00:43.55 | drmessano | Which SIP client? |
| 00:44.12 | drmessano | Stock Linksys firmware? |
| 00:44.34 | Steve_J-obs | yes, it does, I get this message: actually, it registers for a few seconds, and then it says "unable to find key 'xxx' in family sip/registry |
| 00:44.53 | [TK]D-Fender | hopes to see this course birdie'd |
| 00:45.37 | Steve_J-obs | then, I put a stun server on it... stun.fwdnet.net, and it registered... but it would not make the call |
| 00:45.55 | Steve_J-obs | linksys firmware?...one sed |
| 00:46.07 | [TK]D-Fender | thinks "nope... looking like par" |
| 00:46.07 | drmessano | Stock Linksys firmware? |
| 00:46.45 | Steve_J-obs | it is happening with a couple and sip clients, the one I am doing the testing with is XLite |
| 00:47.14 | Steve_J-obs | but then, my linksys-942 phone registers |
| 00:47.30 | drmessano | Are you running the stock linksys firmware? |
| 00:47.32 | MikeJ | sounds like alg on the routher |
| 00:47.52 | Steve_J-obs | 1.00.00 B17 Jun. 25, 2008 linksys wrt610n |
| 00:48.46 | CGMChris | How do you pronounce IAX? eye-axe? e-axe? |
| 00:48.53 | v4mp | guys how would i go about changing this http://v4mpire.pastebin.com/d342ace76 so it sends the queued calls to the agent group i have found its not SIP/1 so wht would i put there so it calls the agent group ? the agent group name is 1 |
| 00:48.54 | drmessano | EEKS |
| 00:48.59 | v4mp | the agents aren't loging in to server as an agent they are logging in as user then using agentcallbacklogin to login as an agent with a diff user to the user to login to server |
| 00:49.20 | CGMChris | takes a shot of jack and EEKS |
| 00:49.26 | v4mp | even a link will do to help but i couldn't find out about it on the book |
| 00:49.37 | bpgoldsb | Asterisk 1.2 'Voicemail(u123)' should be 'Voicemail(123|u);' in 1.6/AEL, right? |
| 00:49.43 | *** part/#asterisk knarfly (n=vtserije@c-75-74-155-198.hsd1.fl.comcast.net) |
| 00:50.00 | CGMChris | exten => s,n(unavail),VoiceMail(${MACRO_EXTEN}@default,u) |
| 00:50.26 | CGMChris | Pipe or comma, pick your favorite.. however commas cause additional overhead as Asterisk converts all commas to pipes prior to parsing. |
| 00:50.41 | hardwire | hmmphmhpmasdf |
| 00:50.45 | [TK]D-Fender | CGMChris: | is gone in 1.6 |
| 00:50.46 | Qwell | CGMChris: incorrect |
| 00:50.56 | CGMChris | ah, i got my info from the book. |
| 00:51.00 | *** join/#asterisk Staos (n=Staosg@pool-72-95-4-187.hrbgpa.east.verizon.net) |
| 00:51.13 | [TK]D-Fender | CGMChris: Just forget that and use "," in place of "|" |
| 00:51.19 | CGMChris | I have been. :-) |
| 00:51.36 | bpgoldsb | [TK]D-Fender: Thats only in the case of Voicemail(), right? |
| 00:51.46 | bpgoldsb | iftime still uses pipes, for example |
| 00:52.16 | [TK]D-Fender | bpgoldsb: Thats a special app. Always read your version's instructions. |
| 00:52.42 | CGMChris | In a few more days I may very well be ready to dive into some Asterisk C code (Now that's a scary thought isnt it?) |
| 00:53.10 | sfire | yes |
| 00:53.25 | [TK]D-Fender | CGMChris: What are you hoping to accomplish? |
| 00:53.28 | drmessano | I know I am scared |
| 00:53.29 | hardwire | do sip sessions renegotiate codecs ever? |
| 00:53.33 | hardwire | during a session |
| 00:53.38 | hardwire | like after ivr |
| 00:53.38 | Steve_J-obs | what I can't figure out, is what would be wrong with this router |
| 00:54.03 | drmessano | Steve_J-obs: Try google.. no one here knows anything about that router |
| 00:54.13 | *** part/#asterisk Staos (n=Staosg@pool-72-95-4-187.hrbgpa.east.verizon.net) |
| 00:54.14 | sfire | which router??? past my scroll back |
| 00:54.19 | CGMChris | D-Fender: My end goal is to produce a plugin module for Asterisk that will allow my clients to plug their "Union Membership Tracking Software" databases into a touch tone system. |
| 00:54.34 | drmessano | Some Linksys BR549-N something |
| 00:54.41 | CGMChris | D-Fender: My much shorter term goal, is to get the system working in-house. :-) |
| 00:54.55 | [TK]D-Fender | CGMChris: plug in to do what? How? |
| 00:55.27 | CGMChris | D-Fender: I sell database systems to Unions (Plumbers, Pipefitters, etc). I want them to be able to have members call in to check balances, make payments, etc. |
| 00:55.54 | CGMChris | D-Fender: All that information is already tracked by the systems I sell... I just need UC. |
| 00:56.19 | [TK]D-Fender | CGMChris: So far nothing that require's C or * coding at all |
| 00:56.41 | CGMChris | D-Fender: Cool, thats good to hear. |
| 00:56.51 | [TK]D-Fender | CGMChris: Go read up on AGI |
| 00:56.56 | v4mp | tk know of any links or how i could change the line where the calls get sent to while in queue to use an agent group ? |
| 00:57.43 | CGMChris | I'll do that right after I figure out this agentlogin/callback deal. Still havent gotten my call queue working properly. |
| 00:58.26 | v4mp | :o |
| 00:59.07 | drmessano | Hey Steve_J-obs |
| 00:59.13 | drmessano | I have a question |
| 00:59.24 | CGMChris | At the moment, it's perplexing that when a phone triggers AgentLogin, it is given hold music... whereas I could click the "login" link on the GUI and it would log that device in...but don't tell me, I'll read first and ask questions later. |
| 00:59.32 | [TK]D-Fender | v4mp: An agent group is a bunch of agents. Each one logs in on their own and how they do so doesn't have to be via the same means. |
| 01:00.19 | v4mp | tk yes but would prefer to run the queued calls into an agent group instead of adding them 1 by 1 into the list |
| 01:00.40 | [TK]D-Fender | v4mp: v4mp Agents still have to log in. |
| 01:01.08 | [TK]D-Fender | v4mp: the only thinkg this saves you is a little organization in queues.conf |
| 01:01.19 | [TK]D-Fender | v4mp: Every agent is every bit as independant as the other. |
| 01:02.16 | v4mp | yea but before i had the queue send the call via the agent to the softphone i was using as (SIP/V4mpire) but would rather change that so it calls the group |
| 01:02.23 | *** join/#asterisk knarfly (n=vtserije@c-75-74-155-198.hsd1.fl.comcast.net) |
| 01:02.36 | *** part/#asterisk knarfly (n=vtserije@c-75-74-155-198.hsd1.fl.comcast.net) |
| 01:03.45 | [TK]D-Fender | v4mp: What group? I don't see a group anywhere.... |
| 01:03.54 | drmessano | Is pastebin down? |
| 01:04.03 | jaytee | oh, god. are we doing the queue thing again? |
| 01:04.14 | drmessano | jaytee: No |
| 01:04.16 | drmessano | jaytee: Yes |
| 01:04.22 | drmessano | jaytee: I dunno |
| 01:04.27 | [TK]D-Fender | jaytee: A possible maybe! |
| 01:04.29 | drmessano | jaytee: May I describe my config? |
| 01:04.36 | [TK]D-Fender | ~iwmwb |
| 01:04.37 | jbot | I WANT MY WEEKEND BACK! |
| 01:04.42 | drmessano | jaytee: There's numbers, some arrows, and some linux |
| 01:04.49 | jaytee | "my fellow prisoners" |
| 01:04.50 | drmessano | jaytee: and a SIP/something |
| 01:04.52 | v4mp | jaytee, im, just trying to get it setup to call an agent group now instead of single user/agent |
| 01:04.57 | [TK]D-Fender | jaytee: Hilarity |
| 01:05.03 | drmessano | DAMNIT |
| 01:05.09 | drmessano | LEARN TO USE A, COMMA |
| 01:05.18 | drmessano | God I hate that |
| 01:05.20 | [TK]D-Fender | v4mp: Show us this group of agents and where you expect them to log in |
| 01:05.28 | v4mp | ok |
| 01:05.34 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:05.50 | jaytee | I was reading and googling today thinking I might setup FOP but then I thought to myself, "Screw FOP! I'm a Dapper Dan man!" |
| 01:06.01 | drmessano | lol |
| 01:06.21 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 01:06.25 | drmessano | FOP really sucks.. to the point that FreePBX now has a config option to disable it in the web interface |
| 01:06.31 | drmessano | It also doesnt work with 1.6 |
| 01:06.39 | v4mp | http://v4mpire.pastebin.com/d18391b63 |
| 01:07.15 | drmessano | jaytee, can you help me with queues? |
| 01:07.20 | [TK]D-Fender | v4mp: How many agents will you have? |
| 01:07.23 | drmessano | Here is what I have so far: |
| 01:07.27 | drmessano | [general] |
| 01:07.32 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 01:07.32 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:07.41 | jaytee | so now I'm looking at these C# .NET libraries for Asterisk AMI and thinking I could design my own replacement for FOP using .NET and .asp and call it Dapper Dan 4 Asterisk, make it GPL2 and sit back and wait for the Cohen brothers to sue me. |
| 01:07.44 | v4mp | tk at present only 2 |
| 01:07.48 | hardwire | any idea how to make spa-9xx's actually hang up? |
| 01:07.49 | drmessano | So now, what would you change to make that work? |
| 01:08.03 | drmessano | hardwire: Put the talk-handle thingie on the button thing |
| 01:08.04 | [TK]D-Fender | v4mp: Screw groups. Total waste. Just make them both members of the queue. |
| 01:08.14 | hardwire | drmessano: ffs.. it's that easy?! |
| 01:08.21 | [TK]D-Fender | hardwire: Write your own firmware |
| 01:08.30 | hardwire | [TK]D-Fender: i'm on it! |
| 01:08.37 | v4mp | drmessano, you really do like to go OTT done you |
| 01:08.38 | drmessano | Mine seem to hang up that way |
| 01:08.46 | drmessano | ffs |
| 01:08.49 | drmessano | "don't" |
| 01:08.52 | [TK]D-Fender | Zomg |
| 01:09.35 | CGMChris | jaytee: I am *still* doing the "queue thing", but WE are not doing it again. I am doing my best to figure it all out. |
| 01:09.37 | v4mp | tk you of a link on how i could learn more abot directing them to multiple agents rather than to the login they use to login to server ? |
| 01:09.58 | drmessano | and for the record, I don't arm wrestle |
| 01:10.05 | drmessano | So stop with the OTT crap |
| 01:10.24 | [TK]D-Fender | v4mp: that looked like english... run through a blender with sprinkles on top |
| 01:10.39 | v4mp | i couldn't find it in the book unless im looking in the wrong place |
| 01:10.53 | [TK]D-Fender | v4mp: Do you even really WANT a queue? What exactly are you looking to have happen? |
| 01:11.14 | drmessano | [TK]D-Fender: Whats the asterisk command for OutSource? |
| 01:11.14 | v4mp | drmessano, what has arm wrestling got to do with anything ? |
| 01:11.34 | [TK]D-Fender | v4mp: I am really beginning to doubt you're even piking the right too. |
| 01:11.50 | [TK]D-Fender | picking* |
| 01:11.51 | v4mp | tk yes for when theres more calls than the agents can take as already talking to someone |
| 01:12.10 | v4mp | picking the right too ? what o_O |
| 01:12.17 | drmessano | v4mp: Over The Top was a 1987 movie starring Sylvester Stallone where he played a truck driver/arm wrestler who was badass enough to drive an 18 wheeler into a mansion. You're NOT that cool.. so stop name dropping, pls/ |
| 01:12.27 | [TK]D-Fender | v4mp: Agents are meant to be MOBILE. You log in because I may be agent 5000, but I'm not sitting at the same DESK as yesterday so I need THIS phone to ring when I'm called. |
| 01:12.43 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 01:12.51 | [TK]D-Fender | v4mp: thats the entire point to agents. Tracking an identity separate from WHERE they are located |
| 01:13.11 | [TK]D-Fender | v4mp: So you need to log in so it knows where to send calls |
| 01:13.27 | CGMChris | D-Fender: It would be desirable at my location to not track who did what, but rather ALWAYS keep all phones logged in as agents so they ring when a call comes in. |
| 01:13.33 | v4mp | drmessano, its also a shortened text speak meaning your going overboard with what your saying |
| 01:14.12 | v4mp | tk well i couldn't find anything mentioning that i could only find things about using agents for queues |
| 01:14.14 | drmessano | you're |
| 01:14.17 | drmessano | Not "your" |
| 01:14.20 | drmessano | or "you are" |
| 01:14.37 | v4mp | true but not important |
| 01:14.44 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-ec7e92ade99bf4a0) |
| 01:14.53 | v4mp | you know what it meant so thats all that matters |
| 01:14.53 | drmessano | Yes, not looking ignorant is overrated |
| 01:15.02 | [TK]D-Fender | v4mp: Who should be a member of this queue? |
| 01:15.16 | [TK]D-Fender | v4mp: is it fair to say the DEVICES that should ring are FIXED? |
| 01:15.22 | jaytee | sprelling and agrammer is highest overrate |
| 01:16.06 | v4mp | fixed being as will be used from the same place ? or same client app/hardware ? |
| 01:16.12 | *** join/#asterisk outtolunc (n=me@c-24-130-75-122.hsd1.ca.comcast.net) |
| 01:16.24 | [TK]D-Fender | v4mp: the same DEVICES are members |
| 01:16.27 | CGMChris | v4mp: that would depend on if the users are arbitrarily moving their phones |
| 01:16.55 | [TK]D-Fender | v4mp: Agents are for when you have 100 iditiot tech support guys working in 3 shifts 24/7 with you only having 10 phones. |
| 01:16.59 | v4mp | tk yes same members and maybe more to add at a later date |
| 01:17.26 | [TK]D-Fender | v4mp: and you want to know how many calls Achmed took. You can't do that based on the phone he was sitting in front of.... |
| 01:17.53 | v4mp | what ? :S |
| 01:18.00 | hardwire | hmm.. I can't get two sip phones to redirect to eachother after some IVR |
| 01:18.31 | [TK]D-Fender | v4mp: It was a simple analogy. I have 100 people working in a call center. they don't each have a phone and they aren't in the office at the same time. Follow me so far? |
| 01:18.35 | hardwire | local sip phone 1 calls ivr.. it dials whatever I dial, another sip phone in the office.. no sip reinvite |
| 01:18.47 | v4mp | yea i get ya |
| 01:19.01 | bpgoldsb | So I think I've got a better understanding of Macro/GoSub now. |
| 01:19.39 | v4mp | each member to answer calls will be in different locations |
| 01:19.42 | bpgoldsb | But is it normal to be calling macro in AEL style '&foo(bar)' without an arg? i.e. 'foo();' |
| 01:19.45 | [TK]D-Fender | v4mp: Well if I have 10 phones I need * to know that employee 74 is "logged in" and can be reached by calling "100@justdialmyphones" in the dialplan |
| 01:20.07 | [TK]D-Fender | v4mp: Not physical locations...... |
| 01:20.21 | drmessano | devices Vs users |
| 01:21.05 | v4mp | each device will be located in there own locations which wont change only thing that will change is when theres another device to add |
| 01:21.43 | CGMChris | D-Fender: My queue issue is *resolved*. Any idea how to get arround the annoying hold music when calling AgentLogin? |
| 01:21.55 | [TK]D-Fender | v4mp: Lets say we're both agents. You feel like slacking off so you log out and just sit at your desk. I walk in and see you like the lazy ass you are and throw you off the roof. And then for convenience I log into the queue with my agent number telling the queue to send the calls to YOUR PHONE |
| 01:22.29 | [TK]D-Fender | CGMChris: you'll sit in silence otherwise |
| 01:22.31 | *** join/#asterisk msalaque (n=m@60-242-151-247.static.tpgi.com.au) |
| 01:22.37 | [TK]D-Fender | CGMChris: for which I suspect you could set the MoH class to a dead-end before entereing |
| 01:22.42 | CGMChris | D-Fender: I would much prefer silence (aka concentration) to the hold music |
| 01:23.25 | [TK]D-Fender | CGMChris: I only know 1 other person who uses AgentLogin. Its horrific and the only reason I'd use it is if the company were a bunch of lazy cheapskates without even real phones. |
| 01:23.36 | hardwire | yay.. sip reinvite worked |
| 01:23.40 | v4mp | ahh i see tk well i guess agents aren't important as the caller wont know who they are being past to until that person answers the call and says there name anyway |
| 01:23.41 | hardwire | diabled tT and wW on dial |
| 01:23.50 | hardwire | reads dem docs gewd |
| 01:24.03 | [TK]D-Fender | v4mp: You are WAY off the track. |
| 01:24.35 | v4mp | how so ? |
| 01:24.35 | [TK]D-Fender | v4mp: this is not for the caller's benifit. |
| 01:24.38 | CGMChris | D-Fender: I don't need a phone system to enforce productivity and limit lazyness, we have 5 employees. I know when people are working. |
| 01:24.39 | v4mp | i didn't say it was |
| 01:25.02 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 01:25.02 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:25.09 | [TK]D-Fender | v4mp: "won't know who they are being past" <-- HUH!??!?! |
| 01:25.51 | v4mp | meaning they aren't going to know who the call is being past to only the department so then agents aren't important |
| 01:26.09 | [TK]D-Fender | CGMChris: Ageng login forces them to either wear a headset with some idle sond at all times and to sit in a call that could get hung up by accident in SO many ways. |
| 01:26.34 | jaytee | speaking of queues, I've setup a test queue and when I use AgentLogin to login and then I hang up it immediately logs me out again. When I use AgentCallbackLogin I stay logged in after I hang up and calls to the queue will ring to my phone. I can't find anything in the docs that explains this behavior. |
| 01:26.43 | [TK]D-Fender | v4mp: the calls gets passed around the group of logged in members of that queue |
| 01:26.56 | v4mp | yea |
| 01:27.05 | [TK]D-Fender | jaytee: This is what I'm explaining to CGMChris here |
| 01:27.05 | jaytee | damn, [TK]D-Fender answered my question right before I finished typing it. |
| 01:27.26 | [TK]D-Fender | jaytee: AgentLogin means you have to SIT in that stupid call until it BEEPS at you and you press a key to answer the call. |
| 01:27.44 | jaytee | [TK]D-Fender, dude, you're psychic! what's tonight Powerball numbers? I'll split the jackpot with you |
| 01:27.51 | [TK]D-Fender | jaytee: Unlike the rest of the unverse that just wants their phone's to ring like the rest of us |
| 01:28.11 | CGMChris | Ok, thats what I want. Calls continue after AgentCallbackLogin... I will use it. Thank you jaytee and D-Fender. |
| 01:28.16 | [TK]D-Fender | AgentLogin = Who really cares? 1% of us |
| 01:28.20 | *** join/#asterisk Goldfisch (n=gregturn@user-142gbp1.cable.mindspring.com) |
| 01:28.40 | jaytee | [TK]D-Fender, that was kind of what I was thinking it was but couldn't find any confirmation in the docs or the book that made that clear. Thanks for clarifying that. |
| 01:28.47 | *** part/#asterisk Goldfisch (n=gregturn@user-142gbp1.cable.mindspring.com) |
| 01:28.57 | [TK]D-Fender | jaytee: the app's instruction kinda allude to that... |
| 01:29.20 | [TK]D-Fender | jaytee: this IS one of those chapters I'd love to re-write. And there are several.... |
| 01:29.32 | jaytee | but I read where AgentCallbackLogin is being deprecated in favor of AddQueueMember and RemoveQueueMember. |
| 01:29.33 | drmessano | Yeah, dumas |
| 01:29.40 | v4mp | but tk only way i can see to add members to the queue is add members as member => Agent/number |
| 01:29.45 | CGMChris | I think I am at a near lethal combination of Jack Daniels, Coffee, Staring at the screen, and sleep deprivation, so I am going to relax and check into that function tomorrow :-) |
| 01:30.49 | CGMChris | v4mp: member => Agent/number <- works for me |
| 01:31.04 | CGMChris | v4mp: You still have to tell the system that the agent is AVAILABLE |
| 01:31.06 | hardwire | wow.. I mostly talk to myself in this channel. |
| 01:32.06 | drmessano | Anyone seen hardwire lately? |
| 01:32.13 | jaytee | hardwire who? |
| 01:32.15 | CGMChris | hardwire: You can take my place and talk to v4mp if you like. |
| 01:32.22 | [TK]D-Fender | v4mp: go read the samples again |
| 01:32.23 | drmessano | HAW |
| 01:32.49 | drmessano | v4mp: Hardwire wrote the queue code for asterisk.. he can help you |
| 01:32.58 | CGMChris | LOL |
| 01:33.11 | russellb | yes, we can blame hardwire ! |
| 01:33.43 | drmessano | v4mp: I can PM you his PM address if you like |
| 01:33.46 | hardwire | v4mp: I know.. rtfm |
| 01:33.59 | CGMChris | hardwire: If only....agents voicemail were ignored while a caller is in a queue, oh, joy...how simple life would be. the GUI would WORK! |
| 01:34.04 | hardwire | not to be rude.. but I'm guessing you're missing the 90% perspiration part :) |
| 01:34.15 | hardwire | and I have no idea what we are talking about |
| 01:34.18 | hardwire | woot! |
| 01:34.36 | drmessano | hardwire: Are you saying v4mp is expecting everyone to do the work for him? How dare you.... TRUTHMONGER |
| 01:34.46 | hardwire | I sells you truths. |
| 01:34.50 | hardwire | then I sells my brother some truths. |
| 01:35.01 | hardwire | then my brother sells you the same truths again.. double price. |
| 01:35.02 | drmessano | Why dont you go take your truths and run for president! |
| 01:35.20 | drmessano | Asterisk is about LIES |
| 01:35.23 | drmessano | and IAX2 |
| 01:35.24 | CGMChris | ok, ok... somebody ask an * question. This is getting weird. |
| 01:35.52 | jaytee | god, everytime I load www.tvguide.com/listings Firefox spikes my CPU into the 90's and then goes unresponsive. just that site. too much flash crap |
| 01:36.08 | hardwire | jaytee: noscript |
| 01:36.11 | drmessano | Firefox has become total crap |
| 01:36.14 | CGMChris | jaytee: that went away when I upgraded from my 500mhz |
| 01:36.19 | drmessano | FF3 is crapware |
| 01:36.35 | jaytee | hardwire, I'm running noscript and I even tried disabling that but still no joy |
| 01:36.47 | drmessano | It doesnt use 2GB of RAM anymore, it just buckles under the slowness of attempted memory management now |
| 01:36.49 | jaytee | drmessano, this is still FF2 |
| 01:36.52 | drmessano | yay 4 progress |
| 01:37.04 | jaytee | I'm still running Ubuntu 7.10 |
| 01:37.19 | drmessano | Screw it, I am going back to Mosaic |
| 01:37.34 | jaytee | ah, those were the days. |
| 01:37.48 | CGMChris | eh, (FF3 / 10) > IE7, on 500mhz winxp w/ 256mb ram. |
| 01:37.50 | jaytee | that wonderful scrolling marquee |
| 01:37.56 | drmessano | Netscape 4.7 was the last awesome browser |
| 01:38.09 | v4mp | tk i did and can only see to do it via agent member not a group and couldn't find in book about the line for extensions |
| 01:38.28 | CGMChris | I miss the animated netscape stars/night sky icon back in win98 <3 |
| 01:38.40 | drmessano | [TK]D-Fender: halp halp |
| 01:39.37 | jaytee | I miss Microsoft Bob. He and I got to be good friends but then we had a falling out over me using AOL instead of MSN and now he doesn't call and won't even write. |
| 01:39.44 | drmessano | lol |
| 01:39.51 | *** join/#asterisk Corydon76-dig (i=one@pdpc/supporter/bronze/Corydon76-home) |
| 01:39.51 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 01:40.19 | drmessano | [TK]D-Fender: I just set up a static agent and it blew up my TDM card. Is that covered under warranty? |
| 01:40.46 | jaytee | I love that one joke showing a Word document and Clippy going, "You appear to be composing a suicide note. Would you like some help with that?" |
| 01:42.51 | *** join/#asterisk talntid (n=t@66.208.251.170) |
| 01:47.57 | v4mp | heh samples have just wasted my time so far and nothing about what im trying to do it book i went through all relevent parts so how about something else more useful the i woldn't b asking for help |
| 01:49.53 | drmessano | Its not a matter of asking for help, its the total lack of self enpowerment. |
| 01:49.54 | [TK]D-Fender | v4mp: And when the samples show you more and I know they do and you don't follow my advice it is my time that is wasted. I don't send people to look at things for no reason. |
| 01:50.09 | jaytee | I'm watching the Switchvox video on Digium's website. The girl in it is kinda hot. Nice cleavage but a bit wide in the hips |
| 01:50.18 | drmessano | Hmm |
| 01:50.20 | drmessano | link pls |
| 01:50.33 | [TK]D-Fender | drmessano: Thats...... right. And not "harsh". I might say almost "deep". I'm impressed |
| 01:50.33 | jaytee | http://www.digium.com/en/products/switchvox/ |
| 01:50.55 | talntid | for sale: rhino r1/t1 card with echo cancellation. $425 shipped. 3 months old. bought new from voipsupply. |
| 01:51.28 | jaytee | rhino pimp! |
| 01:51.38 | talntid | :P |
| 01:51.51 | talntid | but her up, she's makes a good xhristmas gift for the kids. |
| 01:51.57 | hardwire | I'm seeing the world in a whole new light now |
| 01:52.04 | hardwire | asterisk is a sip device. |
| 01:52.13 | russellb | jaytee: she's one of the switchvox founders |
| 01:52.33 | [TK]D-Fender | russellb: Would I be too far off base in saying "I'd hit that"? :) |
| 01:52.34 | jaytee | russellb, no kidding? |
| 01:52.41 | talntid | link to video? |
| 01:52.51 | jaytee | http://www.digium.com/en/products/switchvox/ |
| 01:52.52 | [TK]D-Fender | Link to her address? |
| 01:52.54 | russellb | and married to another one of the founders ;) |
| 01:52.54 | drmessano | [TK]D-Fender: The enjoyable part of any hobby (which I think most of us have to consider some part of our involvement just that, or else we wouldn't be here until all-hours), is finding people you can learn from, share info with, and in turn build off of each others learning and sense of enpowerment.. when it's nothing but leeching, it makes it not-fun. |
| 01:52.56 | talntid | oh, im an idior |
| 01:52.58 | talntid | idiot |
| 01:53.14 | russellb | jaytee: yep ... she's now the Switchvox Product Manager for Digium |
| 01:53.23 | [TK]D-Fender | russellb: remember the Hollywood vows : to love, homour, and cherish.... until something ebtter comes along! |
| 01:53.37 | [TK]D-Fender | Honour* |
| 01:53.44 | [TK]D-Fender | better* |
| 01:53.49 | [TK]D-Fender | DAMN I can't type today... |
| 01:53.54 | jaytee | so is Switchvox just Asterisk with a bunch of extra stuff glommed on or is it a whole different core? |
| 01:53.54 | bpgoldsb | If I want to say "do a while there is a call on the line, do b when the call is hungup', what should I be using? |
| 01:53.57 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 01:54.08 | russellb | jaytee: switchvox uses asterisk. |
| 01:54.12 | [TK]D-Fender | jaytee: Trixbox as sold by Digium ;) |
| 01:54.13 | drmessano | bpgoldsb: English? |
| 01:54.20 | russellb | would be silly for Digium to sell a PBX not based on Asterisk, wouldn't it? |
| 01:54.20 | CGMChris | D-Fender, who is that guy... Mavic Beacon? |
| 01:54.24 | CGMChris | Mavis* even |
| 01:54.26 | bpgoldsb | drmessano: Thanks |
| 01:54.36 | [TK]D-Fender | bpgoldsb: * doesn't do stuff in the background of a call. |
| 01:54.41 | [TK]D-Fender | bpgoldsb: Dialplan is linear |
| 01:54.49 | drmessano | I think we just took down digium.com |
| 01:55.04 | bpgoldsb | [TK]D-Fender: Alright, I'll just continue doing it the way I am. |
| 01:55.05 | drmessano | The #asterisk effect |
| 01:55.30 | bpgoldsb | Ext1 calls external shell script, Ext2 calls kill on that script |
| 01:55.47 | CGMChris | drmessano: digium.com = works from here |
| 01:55.57 | drmessano | hrm |
| 01:56.02 | jaytee | [TK]D-Fender, you were joking, right? it's not really a rebranded Trixbox is it? |
| 01:56.10 | drmessano | The server at www.digium.com is taking too long to respond. |
| 01:56.11 | russellb | jaytee: hell no it's not |
| 01:56.39 | [TK]D-Fender | jaytee: Full server with * + GUI.. Licensed and all that too... |
| 01:56.52 | drmessano | Trixbox is a rebranded Grandstream PBX |
| 01:56.54 | [TK]D-Fender | jaytee: Just not FreePBX. They were independant and got bought up |
| 01:57.18 | jaytee | drmessano, haha that makes perfect sense then |
| 01:57.32 | jaytee | BudgeVox |
| 01:57.37 | drmessano | Damnit |
| 01:57.41 | CGMChris | From the screenshots, trixbox looks like it users Asterisk + FreePBX + Proprietary WebGUI |
| 01:57.41 | drmessano | I cant load that Digium link |
| 01:57.55 | drmessano | No |
| 01:58.02 | drmessano | Its not "FreePBX" anymore |
| 01:58.13 | russellb | drmessano: sure, from the marketing side of things |
| 01:58.15 | russellb | clearly, it is. |
| 01:58.28 | jaytee | what do they call it now? CornholioPBX? |
| 01:58.35 | CGMChris | drmessano: Meaning, they modified FreePBX... or, no more freepbx.org? |
| 01:58.50 | De_Mon | CGMChris I thought mavis beacon was a woman... |
| 01:58.52 | voxter | freepbx still exists. fonality/trixbox forked it into their own project |
| 01:58.54 | drmessano | russellb: Its forked and broken, its insulting to the FreePBX team to consider it "FreePBX" now... |
| 01:59.01 | jaytee | a black woman typing teacher |
| 01:59.02 | De_Mon | goes to google |
| 01:59.06 | CGMChris | De_Mon, who knows... i thought mavis beacon could type tho. |
| 01:59.10 | drmessano | regardless of it's recent lineage |
| 01:59.11 | russellb | drmessano: oic |
| 01:59.27 | De_Mon | CGMChris naw, those who can't "do", "teach" |
| 01:59.48 | drmessano | voxter: No one said FreePBX didnt exist |
| 01:59.55 | [TK]D-Fender | De_Mon: those who can't teach, teach physics @ NYU |
| 01:59.56 | voxter | CGMChris asked. |
| 01:59.59 | drmessano | No |
| 02:00.03 | CGMChris | agreed... but, if a person (such a D-Fender) is having a hard time typing, it's proper to say "YOU NEED MAVIS BEACON!" |
| 02:00.10 | v4mp | tk if that was the case i wold be using it more cuz then i would get things done quicker as i prefer to get things over and done with more helpful sites would have helped me finish this setup days ago |
| 02:00.16 | drmessano | [21:58] <CGMChris> From the screenshots, trixbox looks like it users Asterisk + FreePBX + Proprietary WebGUI [21:58] <drmessano> No [21:58] <drmessano> Its not "FreePBX" anymore |
| 02:00.20 | drmessano | thats what was said |
| 02:00.30 | voxter | and then, <CGMChris> drmessano: Meaning, they modified FreePBX... or, no more freepbx.org? |
| 02:00.31 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 02:00.46 | voxter | drmessano: does that really matter??? |
| 02:00.52 | v4mp | unless im looking in wrong place but i dont see how i am |
| 02:00.56 | drmessano | Yes actually, it does |
| 02:00.58 | [TK]D-Fender | v4mp: Using what more? |
| 02:01.13 | v4mp | the guides |
| 02:01.32 | v4mp | but i cant seem to find the stuff relevent to what im trying to setup |
| 02:01.50 | jaytee | [TK]D-Fender's problem isn't that he needs Mavis Beacon, it's that he needs a break from the Asterisk channel. Spending 16-18 hrs a day in here would make anyone's typing get a bit glitchy after awhile. |
| 02:02.09 | CGMChris | jaytee: Agreed... where is bkruse when you need him? |
| 02:02.24 | [TK]D-Fender | v4mp: You'd have seen a little snippet to show you that members are not only agents |
| 02:02.27 | jaytee | lurking in the AsteriskNOW channel no doubt |
| 02:02.44 | drmessano | Yeah... thats the answer.. why bother learning.. Lets just wheel in another Jeeves |
| 02:02.59 | CGMChris | jaytee: No, I've been in there for 8+ hours. Absolutely NO movement whatsoever. Not even a single "Hello" or "Help me" |
| 02:03.17 | [TK]D-Fender | CGMChris: No news is good news |
| 02:03.37 | De_Mon | no pulse means your dead |
| 02:03.40 | jaytee | maybe he's the one married to that woman in the video, in which case I know where I'd be if I was him and it wouldn't be in an IRC channel |
| 02:03.46 | v4mp | tk is this in the book relevent to my problem if so where abouts so i can check if i have been over it |
| 02:03.57 | [TK]D-Fender | De_Mon: See.. and you were wondering if the suffering would ever end! |
| 02:04.16 | drmessano | Yeah, if its not in the book, TK MUST answer for it |
| 02:04.23 | drmessano | wow |
| 02:04.29 | [TK]D-Fender | v4mp: you just never actually pay attention to what I say. |
| 02:05.03 | jaytee | John Lennon once said, "Genius is pain." Myself, I'm just feeling a little achy. |
| 02:05.13 | CGMChris | D-Fender: I'm sure he would pay, but he was taught that * is free! :-) |
| 02:05.44 | v4mp | tk yes i do |
| 02:06.11 | CGMChris | I am leaving... I'm afraid I will have a hard time using "i forgot my cellphone" as an excuse for coming home from work at 9:30 pm |
| 02:06.33 | CGMChris | Goodnight to all! |
| 02:06.40 | jaytee | "Introducing the newest book from O'Reilly Press, Asterisk for people with ADHD, 1st Edition. |
| 02:06.43 | De_Mon | cya chris thanks for the advise |
| 02:06.56 | v4mp | what would i have to benifit by not listening to u tk other than dragging out my setup |
| 02:07.01 | De_Mon | advice! |
| 02:07.06 | drmessano | I already wrote my Asterisk book |
| 02:07.10 | drmessano | Chapter 1 |
| 02:07.23 | drmessano | --> /query [TK]D-Fender |
| 02:07.27 | drmessano | The End |
| 02:07.33 | jaytee | lol |
| 02:07.47 | voxter | i have a new line for you drmessano |
| 02:07.48 | drmessano | PDF to be available shortly |
| 02:08.00 | drmessano | oh? |
| 02:08.01 | voxter | "omg, halp, my astricks is broken" |
| 02:08.24 | drmessano | ~akerisk |
| 02:08.27 | drmessano | ~akerist |
| 02:08.27 | voxter | voxter: repeat that once every 30 seconds until you get an answer |
| 02:08.34 | voxter | ~asstricks |
| 02:08.35 | jbot | [asstricks] #asstricks, the underground gay Asterisk channel. Be afraid, very afraid |
| 02:08.44 | jaytee | hahaha |
| 02:08.59 | [TK]D-Fender | lol |
| 02:09.06 | drmessano | niiiice |
| 02:09.09 | jaytee | "Asterisk, The Uncertain Future of Your Company" |
| 02:09.47 | [TK]D-Fender | "Asterisk, The Futdhjhsd772365@#%^%#" ... NO CARRIER |
| 02:09.53 | drmessano | [TK]D-Fender is gonna go on vacation for two weeks and 12 dotcoms are gonna go under from PBX failures |
| 02:10.11 | jaytee | subtitled, "How to deploy a VOIP PBX with a broken mouse" |
| 02:10.32 | CGMChris | De_Mon: btw, before I go: If your 50k in systems hasnt already been shipped, I would call first thing in the morning to get setup as a partner (and/or cancel the previous unshipped order). Don't give up or be afraid to cancel an order... a few grand in profit is worth it. |
| 02:10.59 | De_Mon | it should be here tomorrow shipped yesterday |
| 02:11.06 | drmessano | <SumLeach> [TK]D-Fender: You here? I need some help. My PBX is down, and my company is going to fold if you dont "help" me fix my PBX!!!!! plz? ;) ^_^ !!!!!!! HELP!!!! No hurry, just need it back up NOW ZOMG IM GONNA GET FIRED!!!! :( |
| 02:11.15 | CGMChris | ah, nevermind. I'm out |
| 02:11.29 | De_Mon | knowing is half the battle go joe! |
| 02:11.33 | *** part/#asterisk CGMChris (n=chris@mail.cgmyes.com) |
| 02:11.45 | *** join/#asterisk luckyaba (n=lucky@ip70-177-7-204.sb.sd.cox.net) |
| 02:12.04 | [TK]D-Fender | drmessano: You taught me the best way I can think of to respond to situations like that... |
| 02:12.23 | drmessano | oh? |
| 02:12.59 | [TK]D-Fender | drmessano: "Really? What company do you work for? Do you think they'll rush to fill your position in immediately at a pay raise to solve this problem?" |
| 02:13.11 | LiNeTuX_Home | query: if you have the polarity wrong on an FXS port, would it matter? |
| 02:13.16 | drmessano | hah |
| 02:13.17 | [TK]D-Fender | hands in a CV |
| 02:13.31 | [TK]D-Fender | drmessano: I listen to you... even when you are being a complete ass ;) |
| 02:13.37 | drmessano | lol |
| 02:13.48 | jaytee | "Why can't you all stop being such arrogant dicks, quit asking me so many questions about my configuration and just solve my problem for me? I downloaded the PDF and I even opened it!" |
| 02:14.12 | [TK]D-Fender | drmessano: Funny thing is you're more like a court jester... occasionally doling out remarkably practical advise few know to take seriously |
| 02:15.01 | jaytee | hey! that's my job! |
| 02:16.09 | drmessano | [TK]D-Fender: When I tell people "you need to shut up, stop asking in IRC and fix it, or someone is going to take your job", people call me an ass.. when I ask them where they work and the fax number for their HR department, they call me an idiot. Still trying to figure out who's correct.. |
| 02:16.35 | De_Mon | drmessano can't they both be right? |
| 02:16.46 | [TK]D-Fender | drmessano: in some cases its just a question of perspective |
| 02:16.51 | drmessano | De_Mon: Guess we'll find out after the interview |
| 02:16.58 | v4mp | heh probz going to be another week til i get thios sorted then |
| 02:17.03 | De_Mon | drmessano wait, they gave you the info? |
| 02:17.06 | *** join/#asterisk BeeBuu (n=beebuu@218.13.67.240) |
| 02:17.33 | drmessano | De_Mon: If that was the case, the question would have been answered already |
| 02:17.46 | De_Mon | you said after the interview |
| 02:17.59 | drmessano | ~humor |
| 02:17.59 | jbot | i heard humor is Q: Why are the streets of Paris lined with trees? A: Because Germans like to march in the shade. |
| 02:18.04 | De_Mon | nevermind, i've already expended enough brain power on this |
| 02:18.18 | jaytee | wonder if jeev ever got his 3 Polycom 330 to reconfigure again so they'd actually work |
| 02:18.28 | BeeBuu | anyone know is there any system like A2Billing? |
| 02:18.38 | drmessano | jaytee: I think his successor did |
| 02:21.11 | jeev | jaytee |
| 02:21.12 | jeev | i did. |
| 02:21.18 | jeev | it was the big sip.cfg i was using that was breaking it. |
| 02:21.28 | jeev | the one you said i should use ;) |
| 02:22.13 | LiNeTuX_Home | I think I should implement something like the new "Mail Goggles" for our phone system. |
| 02:22.27 | v4mp | thought this is meant to be a support channel so far most people have just tryed to take this piss out of me and failed |
| 02:22.27 | LiNeTuX_Home | Make people think before they make phone calls under certain conditions. |
| 02:22.29 | *** join/#asterisk moy (n=moy@189.169.75.3) |
| 02:22.31 | jaytee | jeev, I didn't tell you to use the "big sip.cfg" Polycom does in their whitepaper on provisioning. |
| 02:22.44 | jeev | ;) |
| 02:22.46 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 02:22.46 | jaytee | and I've had no problems following their instructions, mine all work fine |
| 02:22.49 | jeev | i know i know, i'll check it out later |
| 02:22.59 | jeev | i gotta go to london ;) and i havent bought shit or packed |
| 02:23.18 | v4mp | tk has been helpful but you seem to think i ignore what you say when you try help me but i wouldn't benifit from doing so, so obviously i'd want to and do listen |
| 02:23.27 | jaytee | then you better get to it! should be no problem with an Amex Black Card |
| 02:23.33 | jeev | yea i know, i'm goin tomorrow |
| 02:23.36 | jeev | jaytee, amex is gay |
| 02:23.45 | jeev | v4mp, every time tk yells at you, 10 dogs and cats die |
| 02:23.58 | jaytee | and that's why you have the Centurion card, right? |
| 02:24.02 | v4mp | huh ? |
| 02:24.12 | jeev | the card still sucks jaytee |
| 02:24.13 | jeev | amex si gay |
| 02:24.21 | jeev | their cust service is lame |
| 02:26.30 | drmessano | I bet youre considering giving up the black card, eh? |
| 02:26.39 | drmessano | Just "tired of it"? |
| 02:26.56 | [TK]D-Fender | v4mp: You do when you don't look where I tell you to look |
| 02:27.43 | v4mp | how so where haven't i looked that u've told me to ? |
| 02:28.44 | [TK]D-Fender | v4mp: Where did I tell you to for this? |
| 02:29.24 | v4mp | samples.. which i did before i even asked how to set it up with agent groups |
| 02:32.29 | [TK]D-Fender | v4mp: When I tell you to look NOW, then you should look NOW. You aren't looking for the same things and your eyes are closed, just like your mind. |
| 02:32.47 | [TK]D-Fender | v4mp: "No I've already read it all and can't have missed any clues of value" |
| 02:33.03 | [TK]D-Fender | v4mp: If you know it all I'll stop slowing you down. |
| 02:33.16 | [TK]D-Fender | v4mp: Do carry on. |
| 02:33.47 | v4mp | tk i went through the parts that seemed to be to do with what im trying to do... im been trying to find out how for last few days with no success |
| 02:34.27 | jaytee | earwax? oh, wait no that would only interfere with audio. hmmmm, cataracts maybe? |
| 02:34.47 | v4mp | but you say "look now" like it would have randomly changed within a few hours of last looking but like i said i must be looking in the wrong place which i dont see how |
| 02:37.05 | v4mp | well other parts are for zap channels which im not using |
| 02:41.21 | [TK]D-Fender | v4mp: Where you see one kind of channel you can put ANOTHER |
| 02:42.09 | v4mp | but i dunno what the channels come under other than SIP if thats classed as channel but doesn't help for doing the agent group |
| 02:43.04 | voxter | [TK]D-Fender: do any of your polycom users use headsets? |
| 02:43.44 | pcrane | we need an Linksys SPA8000 urgently |
| 02:43.47 | [TK]D-Fender | v4mp: you don't need AGENTS. And you don't understand GROUPS at all and they were never needed in any capacity for this |
| 02:43.55 | pcrane | we're happy to pay courier fees anywhere in the world |
| 02:43.57 | pcrane | can anyone help? |
| 02:44.02 | [TK]D-Fender | voxter: My call center does |
| 02:44.35 | v4mp | tk which you said earlier but you didn't really give me much to go on how to do it without agents but still using the queues |
| 02:45.05 | voxter | [TK]D-Fender: which ones do they use? I have a bunch of people *trying* to use Plantronics S12's, but they complain about a lot of echo/background noise on them. I even tried dicking with em and couldnt get them to sound perfect. |
| 02:45.17 | pcrane | anyone? |
| 02:46.02 | [TK]D-Fender | voxter: I use M22 amps & H261 headsets |
| 02:46.27 | [TK]D-Fender | m4You are currenly making AGENTS as memebers. DON'T |
| 02:49.09 | v4mp | how so ? |
| 02:49.50 | [TK]D-Fender | v4mp: You can set a device to use OTHER than an Agent. |
| 02:50.02 | drmessano | pcrane: Why URGENTLY? |
| 02:51.00 | v4mp | tk you mean when you login as an agent and you pick where to send the call by using an extension in its own context ? |
| 02:51.43 | [TK]D-Fender | v4mp: forget agents. |
| 02:51.57 | pcrane | drmessano: you able to get some? |
| 02:52.05 | drmessano | pcrane: Why URGENTLY? |
| 02:52.36 | pcrane | there's no stock in aus on nz |
| 02:52.42 | drmessano | It's just an 8 port FXS box |
| 02:52.55 | drmessano | Seems easy to find something at least comparable |
| 02:53.28 | pcrane | that works? |
| 02:53.39 | pcrane | it's a misson critical device |
| 02:53.59 | v4mp | tk then i dont know the route to take as the setup im after requires queues and i dont see how to do it without agents so now im even more lost |
| 02:54.31 | De_Mon | hurrrrm |
| 02:54.37 | pcrane | we're trying to replicate a situation where it's talking to a legacy IVR |
| 02:55.01 | jaytee | http://www.voipsupply.com/linksys-spa8000-g1 <----- $251.99 US |
| 02:55.23 | De_Mon | okay so why would a make all in the dahdi source directory say that I don't appear to have the sources for my kernel installed. the headers are correctly linked in /lib/modules/<KVER>/build |
| 02:56.14 | De_Mon | how would I see where the Makefile was failing? |
| 02:57.00 | sfire | De_Mon: no kernel sources (guess) |
| 02:58.02 | [TK]D-Fender | v4mp: Queues have members. They don't have to be agents. The light is on, the wheel is spinning, but I think the hamster is dead. |
| 02:58.11 | pcrane | does anyone have Garret's DDI? |
| 02:58.32 | drmessano | who? |
| 02:58.36 | drmessano | DDI? |
| 02:58.38 | De_Mon | aww what, there's not .config and thats what it wants |
| 02:58.39 | pcrane | DID? |
| 02:58.43 | pcrane | DDI == DID |
| 02:58.54 | drmessano | Who the hell is Garret? |
| 02:59.15 | pcrane | voipsupply guy |
| 02:59.18 | pcrane | aparently |
| 02:59.19 | De_Mon | sfire yeah, I guess it really did mean source ;p |
| 02:59.43 | v4mp | tk so if i understand correctly it would be member => membersname |
| 03:00.27 | [TK]D-Fender | v4mp: Go read the sample files some more. Then move on and read the WIKI a bit |
| 03:00.59 | v4mp | ok i'll look for what i couldn't see several times before :/ |
| 03:01.09 | sfire | De_Mon: it usually does |
| 03:01.21 | [TK]D-Fender | v4mp: and "membersname" is not a real term of any meaning. You don't seem to see what kind of thing follows the " => " |
| 03:01.27 | sfire | De_Mon: kernel sources are needed for anything "advanced" |
| 03:04.55 | [TK]D-Fender | v4mp: Go look where I told you and see what else can go. |
| 03:05.47 | v4mp | ok it seems i got that sorted as its tech/member |
| 03:05.59 | v4mp | so would be SIP/membername |
| 03:06.34 | [TK]D-Fender | v4mp: where do you keep pulling this magical word "membername" from? |
| 03:06.54 | v4mp | as in SIP/V4mpire |
| 03:07.03 | *** join/#asterisk markgreene (n=markgree@c-69-246-176-17.hsd1.al.comcast.net) |
| 03:07.25 | ManxPower | v4mp: Whatever is after SIP/ has better be [somethinginbracketsinsip.conf] |
| 03:07.27 | [TK]D-Fender | v4mp: athe only place the word "member" belongs is in queues.conf. it is not a "thing". |
| 03:07.31 | voxter | any of you guys doing any salesforce integration/screen pop stuff with asterisk? |
| 03:08.05 | [TK]D-Fender | v4mp: SIP/V4mpire is a DEVIE. You add DEVICES as members. AGENTS are devices. Ones where it goes through the dialplan to do whatever you tell it to do. |
| 03:08.33 | [TK]D-Fender | v4mp: AGENTS need to be logged in. fixed devices don't |
| 03:08.49 | v4mp | V4mpire is set as a device and a member |
| 03:08.52 | sfire | voxter: don't you know.. your supposed to repeat all the info you gave in the automated system :p |
| 03:09.02 | voxter | sfire: haha |
| 03:09.05 | ManxPower | v4mp: and that is why you should not name them the same thing |
| 03:09.09 | sfire | voxter: seriously :p |
| 03:09.15 | sfire | haven't you ever called a business :p |
| 03:09.26 | v4mp | ManxPower, i didn't find anything saying otherwise |
| 03:09.42 | ManxPower | v4mp: it's a bad idea |
| 03:09.53 | v4mp | what makes it so bad ? |
| 03:10.00 | [TK]D-Fender | v4mp: "V4ampire" is not a member. |
| 03:10.08 | [TK]D-Fender | v4mp: DEVICES are members. |
| 03:10.12 | ManxPower | People will assign the SIP device ID as the customer account number or, like I do, use the MAC of the device |
| 03:10.23 | drmessano | He doesnt have a member? ouch |
| 03:10.26 | [TK]D-Fender | ManxPower: Greif you don't want...... |
| 03:10.35 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 03:10.48 | [TK]D-Fender | drmessano: Like in that book by that Russian author... |
| 03:10.49 | drmessano | ManxPower: run like hell, save your weekend! |
| 03:10.57 | [TK]D-Fender | drmessano: Ivanna Cutchacockoff |
| 03:11.04 | drmessano | lol |
| 03:11.40 | ManxPower | drmessano: It has been scientifically shown that the attention span in reading documentation is directly related to penis size. The longer your attention span...well..you get the idea. |
| 03:12.20 | drmessano | [23:07] * Steve_J-obs (n=Chris123@wsip-24-234-220-113.lv.lv.cox.net) Quit (Read error: 110 (Connection timed out)) <---- Thats the sound of a Linksys N router being smashed against a wall |
| 03:12.27 | drmessano | :( |
| 03:12.53 | drmessano | oh well, the Airport Extreme is on sale.. You would think Steve Jobs could get a discount |
| 03:13.25 | v4mp | i still dont really see a problem in doing it as both easier for me to understand |
| 03:13.27 | ManxPower | hugs his Cisco 2120XMs |
| 03:13.45 | ManxPower | v4mp: For one thing we can't tell what you are talking about. |
| 03:14.11 | v4mp | ManxPower, maybe because you haven't been following ? |
| 03:14.36 | ManxPower | And in the future when you want to have one call ring on multiple destinations you'll say to yourself "I sure wish I listened to ManxPower and [TK]D-Fender". |
| 03:14.37 | drmessano | Ah, smart move |
| 03:14.41 | [TK]D-Fender | v4mp: What do you mean "both" You're talking like there are 2 things here. There aren't |
| 03:14.56 | v4mp | as a member and a device |
| 03:14.59 | drmessano | ManxPower gives you the time of day, and you diss him.. |
| 03:15.01 | ManxPower | v4mp: I doubt [TK]D-Fender will continue to help you if you don't, hows that? |
| 03:15.13 | drmessano | Welcome to burning both ends of the bridge |
| 03:15.38 | ManxPower | drmessano: Oh, I've been watching v4mp. [TK]D-Fender has warned me off several times. |
| 03:15.46 | v4mp | ManxPower, i am listening but not that i understand much nor from docs |
| 03:16.01 | ManxPower | v4mp: Its something you learn thru experience |
| 03:17.26 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:17.26 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 03:17.57 | ManxPower | lmadsen: What method do you use to name your sip devices in sip.conf? |
| 03:18.17 | lmadsen | ummm.... typically the MAC address of the phone |
| 03:18.39 | ManxPower | v4mp: You have been reading lmadsen's book |
| 03:18.45 | lmadsen | but that's because I abstract the device away from the user in the database, and have an ast_device, ast_extension, and ast_user table, and they are all merged together using id's |
| 03:18.48 | ManxPower | It is one of several authors, actually. |
| 03:18.58 | lmadsen | lies |
| 03:19.01 | lmadsen | don't listen to him |
| 03:19.04 | ManxPower | lmadsen: I don't do any of that and I still use the MAC |
| 03:19.23 | ManxPower | lmadsen: don't look him in the eye whatever you do! |
| 03:19.39 | ManxPower | [TK]D-Fender: what method do you use? |
| 03:19.48 | lmadsen | ManxPower: yep, you can also use the MAC for the [name] part then assign username and extension with the setvar= part |
| 03:19.51 | lmadsen | there are several methods you can use |
| 03:20.26 | drmessano | I do none of that |
| 03:20.31 | drmessano | I used HEX codes |
| 03:20.37 | drmessano | "Magenta, you're fired" |
| 03:20.48 | drmessano | Makes it easier to deal with callcenter turnaround |
| 03:20.48 | lmadsen | :) |
| 03:20.50 | *** join/#asterisk remowylliams (n=Mare@207.65.58.203) |
| 03:20.57 | [TK]D-Fender | ManxPower: I usually number my SIP peers the same as the common extension number I might have lead to ringing them in the dialplan |
| 03:21.08 | drmessano | #000000 <--- Lasted 3 days |
| 03:21.25 | [TK]D-Fender | burns drmessano for witchcraft |
| 03:21.27 | ManxPower | [TK]D-Fender: That's right, I forgot you are one of *those* people. 8-) |
| 03:21.36 | lmadsen | starts shutting down his servers so he can move them to his desk |
| 03:21.41 | lmadsen | network gonna be down for a couple hours |
| 03:21.55 | drmessano | lmadsen: remodeling the bathroom? |
| 03:22.03 | drmessano | lmadsen: Sorry, thats v4mp |
| 03:22.11 | lmadsen | nope |
| 03:22.18 | lmadsen | moving the servers from one side of the room to the other |
| 03:22.32 | remowylliams | Hello everyone, lost and confused, I've been looking at voip-info.org and can't find the server software choices to decide on. is asterisk still around, where did trixbox come from? etc |
| 03:22.34 | lmadsen | I got a new desk; I no longer need to sit on the couch to work |
| 03:22.46 | lmadsen | asterisk has been dead for several years |
| 03:22.51 | seanbright | is asterisk still around? |
| 03:22.53 | seanbright | you're fired. |
| 03:22.54 | lmadsen | lol |
| 03:22.56 | *** join/#asterisk Steve_J-obs (n=Chris123@ip72-193-106-193.lv.lv.cox.net) |
| 03:22.59 | remowylliams | Drat |
| 03:22.59 | lmadsen | remowylliams: yes -- asterisk is around |
| 03:23.02 | v4mp | so basically in sip.conf it can be named anything i was so [phone1] then just set the username under than ? |
| 03:23.13 | drmessano | remowylliams: Naah, Asterisk hasn't been updated in 7 or 8 years.. we're optimists |
| 03:23.19 | drmessano | MAybe one day!!! |
| 03:23.23 | remowylliams | I was running trixbox as I understood it took over for asterisk@home |
| 03:23.29 | lmadsen | remowylliams: it is the basis software for many projects, like trixbox, fonality, etc... -- they use other software that utilize asterisk as the main system, and then throw other services around it |
| 03:23.30 | seanbright | Duke Asterisk Forever |
| 03:23.35 | seanbright | the new vaporware |
| 03:23.35 | drmessano | ROFL!!! |
| 03:23.50 | remowylliams | Ok so asterisk is the core of trixbox etc. |
| 03:23.58 | De_Mon | Asterisk Nukem Forever |
| 03:24.02 | drmessano | seanbright: Change the locks, i'll call security |
| 03:24.05 | seanbright | Duke Nukem Asterisk |
| 03:24.06 | lmadsen | remowylliams: all these other guys are fooling with you -- asterisk is updated several times a day. It is extremely active in development |
| 03:24.10 | seanbright | there, all the permutations. |
| 03:24.24 | De_Mon | Nuke *Asterisk* Nukem Forever |
| 03:24.25 | drmessano | Nukem Duke, Asterisk |
| 03:24.25 | De_Mon | booya |
| 03:24.34 | seanbright | ~thwack De_Mon |
| 03:24.35 | jbot | ACTION smacks De_Mon on the forehead with a UNIX Manual |
| 03:24.36 | De_Mon | crap duke |
| 03:24.43 | lmadsen | remowylliams: correct; basically you can just use asterisk, and build your own full system, or you can use something like trixbox where you utilize what they provide you. Once you need to do specific things, then you should be moving towards using vanilla asterisk |
| 03:24.53 | drmessano | ~trixbox |
| 03:24.54 | jbot | methinks trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/. We do not recommend using it. |
| 03:25.00 | remowylliams | lmadsen: I kind of figure it is but I'm kind of sick with trixbox pushing for a you to register on their forum etc etc etc. |
| 03:25.03 | De_Mon | wow, that make failed miserably |
| 03:25.08 | lmadsen | remowylliams: think trixbox is like a linux distribution, and asterisk is Linux From Scratch. |
| 03:25.15 | drmessano | trixbox is evil |
| 03:25.27 | drmessano | trixbox kills kittens, babies, and votes for McCain |
| 03:25.28 | remowylliams | lmadsen: I started using asterisk thanks to asterisk@home |
| 03:25.34 | lmadsen | ~docs |
| 03:25.34 | jbot | [docs] for basic documentation of Asterisk ask see http://voip-info.org/ (~voip-info) and TheBook (~book) |
| 03:25.52 | lmadsen | what is the quick link for multiple doc references? |
| 03:26.01 | drmessano | ~google |
| 03:26.01 | jbot | from memory, google is a search engine found at http://www.google.com/ |
| 03:26.09 | v4mp | so then i'd be using member => SIP/phone1 correct ? |
| 03:26.10 | *** kick/#asterisk [drmessano!n=Leif@asterisk/documenteur-extraordinaire/blitzrage] by lmadsen (try being useful please!) |
| 03:26.11 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 03:26.13 | ManxPower | lmadsen: well ~docs WAS that line. |
| 03:26.18 | ManxPower | ~doc |
| 03:26.18 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 03:26.21 | lmadsen | aha |
| 03:26.23 | ManxPower | oh, there it is! |
| 03:26.28 | remowylliams | Nothing wrong with voting for McCain if you really think he'll do a better job, crazy as that may be. It's part of being an american. |
| 03:26.29 | drmessano | !kickban lmadsen |
| 03:26.48 | drmessano | !shitlist lmadsen |
| 03:26.51 | *** kick/#asterisk [lmadsen!n=Leif@asterisk/documenteur-extraordinaire/blitzrage] by lmadsen (stop using your authority for evil!) |
| 03:26.53 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:26.53 | drmessano | Damn, no powers here |
| 03:26.53 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 03:27.06 | *** join/#asterisk LARefugee (n=chatzill@c-76-104-191-194.hsd1.wa.comcast.net) |
| 03:27.18 | ManxPower | Your jbot-fu is weak master drmessano! |
| 03:27.22 | De_Mon | Obama uses trixbox you should too |
| 03:27.29 | seanbright | drmessano: /server !kickban lmadsen |
| 03:27.31 | seanbright | duh! |
| 03:27.56 | *** part/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 03:28.04 | remowylliams | Whatelse is around that can be dropped on a CD, dropped into a computer and presto-ing into a voip server with solid x100p zap goodness? |
| 03:28.08 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 03:28.11 | seanbright | he knows better... he's just playing along |
| 03:28.14 | seanbright | god bless him |
| 03:28.30 | jaytee | good god! could we get any more out of date references to documentation? voip-info.org? handbook-draft.pdf? that things like 5 years out of date. |
| 03:28.35 | lmadsen | "solid x100p goodness" are not the words I would necessarily use. |
| 03:28.48 | drmessano | "Introducing, Asterisk 1.2!" |
| 03:28.51 | drmessano | Yeah, thats new |
| 03:28.58 | lmadsen | remowylliams: www.asterisknow.org |
| 03:29.08 | seanbright | i like the ones like "NEW IN CVS 03/07/2004 - " |
| 03:29.31 | remowylliams | lmadsen: very hard to us anything else and have an fxo |
| 03:29.38 | remowylliams | err to use |
| 03:29.41 | drmessano | Or anything that references 1.2.. like, you know, Sarah Palin seeing dinosaurs from her porch |
| 03:29.51 | drmessano | doncha know |
| 03:30.29 | lmadsen | joe sixpack! |
| 03:30.40 | bkw_ | dog gone it! |
| 03:30.45 | remowylliams | *reads on asterisknow* |
| 03:30.48 | drmessano | lmadsen is very well versed in AsteriskWin32.. maybe he can help get that X100P working in Vista for ya |
| 03:31.27 | jaytee | they should just print a cover sheet on every upgrade.txt file that says, "Warning! the useful application or function that you're custom solution will hinge on will be deprecated and replaced to make room for newer, snazzier and less useful ones in the next release. |
| 03:31.37 | drmessano | "Ok, I just installed this X100P card on my Vista64 machine with AsteriskWin32, and windows sees it as a MODEM! WTF???" |
| 03:31.46 | seanbright | jaytee: no more deprecation |
| 03:32.13 | jaytee | seanbright, and that's straight from the mouth of the Pope? |
| 03:32.13 | lmadsen | seanbright: that is true with the asterisk CLI |
| 03:32.53 | seanbright | lmadsen: from a discussion today on -dev, it seems like it's going to be a more general rule, but i may be mistaken. |
| 03:33.15 | seanbright | jaytee: no, just a cardinal. the pope only gets involved when it comes to the kernel. |
| 03:33.18 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 03:33.41 | seanbright | (and by -dev i mean #asterisk-dev, not the mailing list) |
| 03:33.44 | drmessano | Does that mean Russell is Jesus? |
| 03:33.51 | drmessano | Lovely.. there goes his ego |
| 03:34.03 | De_Mon | wait what? pope > jesus? |
| 03:34.12 | v4mp | right how to i make the members for a queue available if im not using the agent route? |
| 03:34.17 | jaytee | because I'm almost positive I read that AgentCallbackLogin was being deprecated in favor of AddQueueMember and RemoveQueueMember. |
| 03:34.22 | drmessano | v4mp: Trixbox |
| 03:34.45 | seanbright | jaytee: between 1.4 and 1.6? yes. |
| 03:34.48 | v4mp | why would i want to use something else when i nearly have the setup done |
| 03:35.01 | De_Mon | depriciation is good for the soul, keeps me employeed too hahaha |
| 03:35.12 | seanbright | you know what, take everything i am saying with a grain of salt. i'm pretty stoned at the moment. |
| 03:35.34 | jaytee | so I'll just have to rewrite that part of the dialplan when I move to 1.6 as well as all the DAHDI crap. |
| 03:35.52 | seanbright | jaytee: the DAHDI changes are far from invasive. |
| 03:36.00 | [TK]D-Fender | jaytee: He doesn't need logins of any kind.... |
| 03:36.05 | seanbright | jaytee: if it took you more than 10 minutes i'd be surprised. |
| 03:36.19 | seanbright | jaytee: as for the agentcallbacklogin stuff... |
| 03:36.21 | seanbright | shrugs |
| 03:36.31 | drmessano | What about DAHDI and OSLEC? |
| 03:36.34 | drmessano | Anyone? |
| 03:36.41 | jaytee | because I'm sure it's more involved than something like sed 's/Zap/DAHDI/' filein fileout |
| 03:36.49 | De_Mon | the zap dahdi changes sound like something a global search and replace can fix |
| 03:37.30 | De_Mon | jaytee the upgrade doc didn't give me that impression, what did you read? |
| 03:37.59 | seanbright | jaytee: that should be all that you need to do from an extensions.conf standpoint. |
| 03:38.10 | seanbright | drmessano: what about it? |
| 03:38.10 | jaytee | hey, I'm just grousing. I'm sure there are relevant reasons for the changes when deprecations are made and I can adjust. |
| 03:38.20 | remowylliams | Why do I twitch when ever I see the phrase 'relatively inexpensive hardware' |
| 03:38.32 | De_Mon | remowylliams you're broke? |
| 03:38.58 | ManxPower | jaytee: The owner of the trademark of "Zaptel" asked Digium to stop using the name. Everyone talked and Digium renamed it to DAHDI. |
| 03:39.14 | drmessano | I see no mention of OSLEC working with DAHDI yet |
| 03:39.17 | jaytee | ManxPower, I know |
| 03:39.21 | drmessano | no patches |
| 03:39.22 | remowylliams | relative to what? a pack of gum, a 20.00 calling card for a cell phone? the new national debt? : P |
| 03:39.52 | ManxPower | hugs his tellabs hardware echo cancelers |
| 03:40.00 | De_Mon | well, of all those a cell phone is the closest thing to 'hardware' |
| 03:40.06 | seanbright | drmessano: well get to patching, man. |
| 03:40.41 | drmessano | seanbright: Coding is a waste of my time. I'm much more of a dictator |
| 03:40.55 | remowylliams | I don't suppose they got the zaptel drivers working on FreeBSD |
| 03:41.22 | De_Mon | remowylliams think X100P if it's talking about pstn interfaces |
| 03:41.24 | *** join/#asterisk Steve_J-obs (n=Chris123@wsip-24-234-220-113.lv.lv.cox.net) |
| 03:41.29 | jaytee | lmadsen, if you and your buddies ever get around to working on a 3rd edition of the book that focuses on 1.6 I'll gladly volunteer to proofread the drafts. |
| 03:41.56 | drmessano | Steve_J-obs: Im gonna wager to say your router sux |
| 03:42.21 | remowylliams | De_Mon: I have a Wildcat X100P clone card, have had it for about 3 years |
| 03:42.25 | jaytee | Steve_J-obs, I thought you had a heart attack? what are you doin in here? |
| 03:42.29 | seanbright | drmessano: dicktator maybe... |
| 03:42.31 | seanbright | zing! |
| 03:42.39 | Steve_J-obs | drmessano: I found the problem! |
| 03:42.41 | De_Mon | remowylliams it was relitively inexpensive, hope that helps! |
| 03:42.47 | remowylliams | plugs into the back of a pipeline 75 |
| 03:42.52 | v4mp | how do i get the device to show as ready to take calls when the member is logged in ? |
| 03:42.58 | De_Mon | well, it was more like dirt cheap but, whatever. close enough |
| 03:43.29 | seanbright | ok, seriously going to bed now. |
| 03:43.32 | Steve_J-obs | I dmz to the computer where I had x-lite installed and it worked! |
| 03:43.34 | seanbright | kbye. |
| 03:43.36 | remowylliams | De_Mon: yes I got mine for about 24.00 with shipping which is funny the card was 9.95 |
| 03:44.06 | Steve_J-obs | but, here comes the question: how can I make it work for 5 computers in the same router?? |
| 03:44.56 | Steve_J-obs | before anything: HI GUYS |
| 03:45.12 | drmessano | Steve_J-obs: I did too, you keep disconnecting |
| 03:45.22 | jaytee | hahaha, Steven Colbert has a rotary dial phone with a Star of David in the middle of the dialer and the ringer cycles in a way that sounds like Hava Nagila. |
| 03:45.25 | drmessano | What is the problem? |
| 03:45.30 | Steve_J-obs | it is the freaking router |
| 03:45.41 | drmessano | no shit? |
| 03:45.59 | drmessano | I never would have... |
| 03:46.06 | Steve_J-obs | I dont know, but it got solved when I dmz it to the laptop... |
| 03:46.13 | v4mp | heh name device diff doesn't allow member to login ah well |
| 03:46.37 | drmessano | Steve_J-obs: Put a WRT54G behind it, in the DMZ! |
| 03:46.47 | drmessano | Steve_J-obs: S0lv3d |
| 03:47.07 | Steve_J-obs | we have 5 computers each with xlite on it, now I see that it only works if I dmz the computer,but how do I do with the other 4? |
| 03:47.34 | drmessano | Steve_J-obs: This doesnt make sense at all. RouterFAIL. You're doing it wrong |
| 03:47.49 | drmessano | Steve_J-obs: You have something mucked up |
| 03:48.08 | drmessano | What kind of modem is it? |
| 03:48.22 | drmessano | DSL, Cable? |
| 03:48.38 | Steve_J-obs | cable modem, arris, connected to a linksys wrt610 dual band |
| 03:48.49 | Steve_J-obs | I have no issuses with the modem |
| 03:48.58 | drmessano | Does the WRT610 WAN have a public IP? |
| 03:49.15 | drmessano | Steve_J-obs: Stop telling me what you think is working |
| 03:49.17 | Steve_J-obs | it is on bridge mode |
| 03:49.24 | drmessano | bridge mode? |
| 03:49.30 | Steve_J-obs | passthru? |
| 03:49.38 | drmessano | Cable modems generally dont have bridge or passthru |
| 03:49.49 | drmessano | Thats DSL terminology |
| 03:50.12 | drmessano | Business or residential? |
| 03:50.15 | Steve_J-obs | ok, just checlked.. the router has a public ip |
| 03:50.46 | Steve_J-obs | business |
| 03:50.51 | drmessano | It sounds like theres a 1-1 NAT |
| 03:51.00 | Steve_J-obs | yes |
| 03:51.06 | drmessano | Thats your problem |
| 03:51.19 | Steve_J-obs | what's a 1-1 nat? |
| 03:51.34 | drmessano | Put the router in a DMZ on the modem |
| 03:51.36 | jaytee | http://www.paperworketc.com/Products/Life_Graphics/M43%20One%20day%20at%20a%20time%20mag.jpg |
| 03:51.50 | Steve_J-obs | mmm... that makes sense |
| 03:51.56 | drmessano | I do this all the time with Comcast SMC gateways |
| 03:52.03 | drmessano | Youre double NAT'ed |
| 03:52.12 | Steve_J-obs | that's smart, my friend |
| 03:52.22 | *** join/#asterisk metfan2007 (n=jc@189.180.216.169) |
| 03:52.30 | Steve_J-obs | I will try that, thank you |
| 03:52.34 | Steve_J-obs | be back :) |
| 03:54.01 | metfan2007 | Hi all! I'm having some issues regarding generating calls using euroisdn in a PRI link, I'm getting the messages like http://pastebin.ca/1223408 while trying to make a call, but I don't understand what is the problem :( can you help me? |
| 03:54.11 | *** join/#asterisk sluxor (n=sluxor@d58-110-243-84.per6.wa.optusnet.com.au) |
| 03:56.33 | sluxor | Is there a way I can use Asterisk as an h323 server for netmeeting and ekiga clients? If the asterisk server was outside the organization would people in the organization sharing an ADSL2+ connection with NAT all be able to use it at the same time? |
| 03:57.26 | drmessano | Oh lord |
| 03:57.29 | drmessano | Why H323? |
| 03:59.26 | sluxor | well better ideas? |
| 03:59.29 | metfan2007 | I want to know the meaning of the pri debug lines :( |
| 03:59.32 | sluxor | im open to suggestions |
| 03:59.33 | drmessano | SIP? |
| 03:59.35 | ManxPower | metfan2007: The telco is daying that's not a valid number "Unallocated (unassigned) number (1), class = " |
| 03:59.40 | sluxor | SIP with video? |
| 03:59.42 | drmessano | The 90s are over |
| 04:00.12 | metfan2007 | ManxPower: mmmm, carrier problem then... I have to call them... thnks! |
| 04:00.13 | ManxPower | metfan2007: for most stuff you can just leave pri debug off and check the value of HANGUPCAUSE after your Dial line. |
| 04:00.14 | drmessano | Eyebeam works fine |
| 04:02.06 | ManxPower | to learn what hangupcause is what, look here www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf |
| 04:02.33 | ManxPower | metfan2007: usually you want pridialplan= to not be set or be set to unknown |
| 04:03.01 | jaytee | ManxPower has so many good links for stuff |
| 04:03.48 | ManxPower | jaytee: Oh, I have a copy of that PDF on my laptop at all times. I had to find the original location via google |
| 04:04.07 | jaytee | I just saved the PDF. Thanks! |
| 04:04.08 | metfan2007 | ManxPower: let my try the pridialplan |
| 04:04.18 | ManxPower | metfan2007: what is it set to right now? |
| 04:05.02 | metfan2007 | ManxPower: it is unset |
| 04:05.27 | metfan2007 | ManxPower: I'm creating the variable using unknown |
| 04:06.06 | ManxPower | metfan2007: try to make local, toll, and international calls. If one kind works and another kind does not, fine someone that has an EuroPRI and get their pridialplan=, prilocaldialplan=, etc settings. |
| 04:06.46 | metfan2007 | ManxPower: Wow, setting it to unknown works! |
| 04:07.18 | ManxPower | <-- smarter than he looks. feel free to send a donation via paypal to eric@fnords.org |
| 04:08.12 | coil | yes |
| 04:08.14 | coil | he isi |
| 04:08.22 | ManxPower | metfan2007: I think in 1.6 it defaults to unknown |
| 04:08.32 | *** join/#asterisk StephenF (n=stephen@c-67-188-58-4.hsd1.ca.comcast.net) |
| 04:08.51 | metfan2007 | ManxPower: Ok, 'cause I'm using the 1.4 branch |
| 04:20.09 | *** join/#asterisk DigitalIrony (n=eric@nat/digium/x-f7ee458ff415dce7) |
| 04:21.27 | *** join/#asterisk DigitalIrony (n=eric@nat/digium/x-2115fb44267ea5e9) |
| 04:24.45 | metfan2007 | ManxPower: Now I can make calls :) thnks, but I'm seeing that when the called party hangups the line the caller only hear silence instead of the tones or Asterisk hangups the line... is the same about priblahblah config lines? |
| 04:27.22 | ManxPower | metfan2007: it is up to your dialplan to look at the DIALSTATUS or HANGUPCAUSE and do the correct thing, play a busy signal, play a congestion tone, hangup. |
| 04:29.04 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25) |
| 04:31.06 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 04:31.06 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 04:31.25 | lmadsen | w00t! Internets back up |
| 04:32.06 | jeev | too bad |
| 04:32.09 | jeev | i couldn't hold it down long enough1? |
| 04:32.09 | jeev | ! |
| 04:32.46 | lmadsen | o.O |
| 04:32.51 | jeev | hey |
| 04:32.53 | jeev | want to continue our thing |
| 04:32.57 | jeev | what you would do for 500k/year ? |
| 04:33.11 | jeev | would you pick the stomach lint out of tom leykis and 9 other fat guys daily ? with your fingers |
| 04:34.45 | lmadsen | sure |
| 04:34.51 | jeev | really ? |
| 04:35.04 | lmadsen | why not... that's not a very hard job |
| 04:35.07 | lmadsen | and I can wash my hands |
| 04:35.11 | jeev | yea but you can't use anti bacterial |
| 04:35.14 | jeev | you have to use regular soap |
| 04:35.24 | lmadsen | that's fine. anti-bacterial creates super bugs |
| 04:35.26 | jeev | dood you've said yes to everything i've offered |
| 04:35.29 | lmadsen | I don't use anti-bacterial anything |
| 04:35.32 | jeev | really ? |
| 04:35.35 | jeev | super bugs eh |
| 04:35.36 | jeev | true |
| 04:35.41 | jeev | the ones it doesn't kill off, build immunity |
| 04:35.45 | jeev | what the hell WOULDn't you do ? |
| 04:35.56 | lmadsen | kills 99.9% of germs... which means we leave the strongest 0.1% to multiply... sounds like a bad scheme |
| 04:35.59 | jeev | yep |
| 04:36.05 | jeev | that 99.9 % is stuff |
| 04:36.08 | lmadsen | goes back to setting up his home office |
| 04:36.09 | jeev | it aint uptime! |
| 04:36.10 | jeev | no |
| 04:36.12 | jeev | dont leave dood |
| 04:36.14 | jeev | you're the only one here |
| 04:36.15 | jeev | i DEMAND you stay |
| 04:40.07 | drmessano | Pay him off with your AMEX Orgasmitron Card |
| 04:42.52 | *** part/#asterisk remowylliams (n=Mare@207.65.58.203) |
| 04:48.10 | *** join/#asterisk axisys (n=axisys@117.18.231.150) |
| 04:50.22 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 04:54.28 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 04:54.46 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 04:55.34 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.13) |
| 05:04.00 | *** join/#asterisk johnakabean (n=none@pool-72-82-113-223.nrflva.east.verizon.net) |
| 05:04.22 | johnakabean | hey guys what is the easiest way to update asterisk from 1.4.22 or w/e to 1.6 |
| 05:06.18 | Qwell | using zaptel? |
| 05:08.04 | *** join/#asterisk pluesch0r (n=pluesch0@vie-086-059-124-159.dsl.sil.at) |
| 05:08.16 | [TK]D-Fender | johnakabean: Backupt your configs, fluch your modules folder. Install 1.6 over. Copy back relevant configs. Correct for version changes |
| 05:10.09 | lmadsen | one telephone back up |
| 05:15.20 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 05:20.36 | *** join/#asterisk jameswf-home (n=james@ip68-2-99-240.ph.ph.cox.net) |
| 05:20.47 | CrashSys | Anyone ever messed with the GrandStream GXW4024 24-FXS media gateway? |
| 05:21.22 | jameswf-home | ~grandstream |
| 05:21.23 | jbot | grandstream is probably the Yugo of VoIP hardware. Run. Run away now. |
| 05:22.34 | CrashSys | Well I was holding on to the fantasy that maybe a different group of chinese engineers (competent ones) made the gateways as compared to their phones :) |
| 05:22.48 | jameswf-home | actualy all made in taiwan |
| 05:23.06 | CrashSys | sorry, taiwanese engineers |
| 05:23.13 | jameswf-home | same factory as some tdm cards |
| 05:24.09 | CrashSys | Ehh... I was just curious... i'll probably go with AudioCodes or MediaTrix... |
| 05:24.59 | johnakabean | well i just ran make and configure with --prefix=/usr --mandir=/usr/share/man --infodir=/usr/share/info --enable-shared --enable-threads=posix --enable-checking=release --with-system-zlib --enable-__cxa_atexit --disable-libunwind-exceptions --enable-libgcj-multifile --enable-languages=c,c++,objc,obj-c++,java,fortran,ada --enable-java-awt=gtk --disable-dssi --enable-plugin --with-java-home=/usr/lib/jvm/java-1.4.2-gcj-1.4.2.0/ |
| 05:25.04 | johnakabean | is this ok? |
| 05:25.22 | trnzmeta | guys: what would should I be looking at if the monitor recordings keep dropping out after a while |
| 05:25.51 | johnakabean | space available, trnz,??? I have no problem with the recordings unless I change it from wave to gsm or someting else |
| 05:26.32 | trnzmeta | hmm actually, I have a cron script that ssh files across every 5 minutes |
| 05:26.51 | trnzmeta | to save space, maybe it's doing it mid conversation |
| 05:27.07 | trnzmeta | however won't asterisk lock or not write the file until the phone call is finished |
| 05:28.07 | *** part/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 05:32.27 | *** join/#asterisk shinao1 (n=shinao1@83.229.85.138) |
| 05:33.34 | lmadsen | correct, the file would be locked |
| 05:38.23 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 05:38.35 | *** join/#asterisk Carlos_PHX (n=Carlos@vpn.pensus.com) |
| 05:39.56 | *** join/#asterisk pcrane (n=pcrane@125-236-160-52.broadband-telecom.global-gateway.net.nz) |
| 05:40.23 | *** join/#asterisk Carlos_PHX (n=Carlos@237.sub-75-210-228.myvzw.com) |
| 05:42.45 | lmadsen | can someone check that http://tfot.leifmadsen.com works for the outside world? |
| 05:42.56 | lmadsen | feel free to check www and blog as well :) |
| 06:09.19 | jeev | yea leif |
| 06:09.20 | jeev | it works |
| 06:09.42 | lmadsen | sweet |
| 06:09.48 | jeev | i wanna mirror that shit |
| 06:10.39 | jeev | dont make me guess your blog password |
| 06:10.42 | jeev | and put revealing photos |
| 06:11.04 | lmadsen | you won't guess it |
| 06:11.21 | lmadsen | only way to do it would be to brute force it |
| 06:11.28 | jeev | i already got it |
| 06:11.31 | lmadsen | sweet |
| 06:11.38 | yang | Hello Leif ! |
| 06:11.44 | lmadsen | well hello there! |
| 06:11.46 | jeev | i'm gonna paste it here |
| 06:11.50 | lmadsen | pls do |
| 06:12.00 | jeev | "jeev-rules-all" |
| 06:12.12 | lmadsen | omg how'd you know? |
| 06:12.19 | jeev | dood |
| 06:12.21 | jeev | what the hell |
| 06:12.24 | jeev | you talk about yoru new business logo |
| 06:12.26 | jeev | but dont show it |
| 06:12.27 | jeev | that's wack |
| 06:12.38 | yang | took the Asterisk bok to his journey:) |
| 06:12.41 | lmadsen | it's the logo being used |
| 06:12.49 | lmadsen | yang: excellent :) |
| 06:12.54 | lmadsen | I need to setup my speakers |
| 06:12.57 | jeev | yang, you know.. i'm the one who actually wrote the bok |
| 06:12.58 | jeev | book |
| 06:13.00 | jeev | lmadsen just stole it |
| 06:13.04 | jeev | and didn't give me credit |
| 06:13.13 | jeev | and since that happened, i forgot EVERYTHING that i wrote |
| 06:13.22 | jeev | now i continually ask for help so i can punish him |
| 06:14.04 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-94-244.w86-215.abo.wanadoo.fr) |
| 06:15.01 | yang | lmadsen: Does Jared Smith comes arond here too? |
| 06:15.08 | lmadsen | sometimes... but not too often |
| 06:15.18 | lmadsen | he is on irc... just not in this room a ton |
| 06:15.22 | MikeJ | lmadsen: !! |
| 06:15.32 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:15.34 | lmadsen | Jerris? |
| 06:15.39 | MikeJ | indeed |
| 06:16.10 | voxter | hey leif |
| 06:17.05 | voxter | did you ever figure out how to change polycom's config so it didnt use the strange new character-like input mode? |
| 06:17.08 | lmadsen | MikeJ: howdy! |
| 06:17.24 | MikeJ | hi hi.. tell jim to go to sleep |
| 06:17.29 | yang | lmadsen: I think it would be really nice to see some sort of an Asterisk workshop at Linuxtag in Berlin next year :) |
| 06:17.30 | lmadsen | voxter: I just hit the A/a/1 button... |
| 06:17.50 | lmadsen | MikeJ: request received and performed |
| 06:17.57 | MikeJ | hehe |
| 06:18.05 | yang | lmadsen: I couldn't find any other Events in Europe, Astricon usually takes place in Arizona |
| 06:18.20 | voxter | lmadsen: hm, i wonder if there's a default someplace in the cfg file. |
| 06:21.59 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 06:23.10 | MikeJ | yang: wasn't there stuff there? |
| 06:23.19 | MikeJ | there is also fostel |
| 06:23.44 | MikeJ | thats a bit different of a kind of conf but well worht attending |
| 06:23.55 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 06:27.10 | yang | MikeJ: yes, was Asterisk present at Fostel? |
| 06:27.31 | MikeJ | didn't make this last one.. |
| 06:27.47 | yang | ah its a Free Software/Open Source Telephony Summit |
| 06:27.58 | *** join/#asterisk implicit- (n=bayan@unaffiliated/implicit) |
| 06:28.19 | yang | so its gotta be specialised in this subject |
| 06:30.52 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 06:31.13 | yang | MikeJ: Any idea where this year's meeting is gonna be held? |
| 06:31.38 | yang | MikeJ: can't find the infos about 2008/2009 |
| 06:31.50 | MikeJ | none |
| 06:33.08 | MikeJ | I was at http://fostel2007.fostel.org/index.php/Main_Page |
| 06:35.10 | MikeJ | looks like its a conf in need of a good sponsor |
| 06:44.39 | *** join/#asterisk jeffspeff (n=jeff@c-69-245-31-86.hsd1.ky.comcast.net) |
| 06:46.41 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 06:46.54 | drmessano^ | ~assterisk |
| 06:46.59 | drmessano^ | Hmm |
| 06:50.28 | voxter | ~asstricks |
| 06:50.30 | jbot | i heard asstricks is #asstricks, the underground gay Asterisk channel. Be afraid, very afraid |
| 06:50.31 | voxter | that what you're looking for? |
| 06:51.13 | *** join/#asterisk Chris-NB (n=chris@nfw.ecos.at) |
| 06:53.03 | drmessano^ | no clue |
| 06:53.07 | drmessano^ | Im half asleep |
| 06:54.52 | coil | lol |
| 06:54.58 | drmessano^ | So when is IAX2 being renamed? |
| 06:57.10 | voxter | i wonder if it will be renamed to MAHMI |
| 06:58.20 | drmessano^ | nah |
| 06:58.33 | drmessano^ | D _ _ D _ |
| 07:00.56 | drmessano^ | Actually |
| 07:01.01 | drmessano^ | D _ _ D I |
| 07:02.22 | MikeJ | drmessano^: is there an h? |
| 07:02.37 | drmessano^ | lol |
| 07:02.55 | voxter | id like to buy a vowel |
| 07:03.04 | MikeJ | Y |
| 07:03.17 | drmessano^ | Reminds me of family guy |
| 07:04.11 | drmessano^ | I'll take a Z |
| 07:04.14 | drmessano^ | The number 4 |
| 07:04.32 | drmessano^ | Q |
| 07:04.35 | drmessano^ | Another Q |
| 07:04.37 | drmessano^ | A third Q |
| 07:04.42 | drmessano^ | and the Batman symbol |
| 07:06.07 | msalaque | <PROTECTED> |
| 07:06.34 | drmessano^ | slaps the everloving crap out of msalaque |
| 07:06.40 | drmessano^ | Did you not see that not work? |
| 07:09.03 | msalaque | drmessano^: i am new in here . so just checking |
| 07:09.09 | *** join/#asterisk jplank (n=GBove@ool-18bb018e.dyn.optonline.net) |
| 07:09.29 | *** join/#asterisk Chris-NB (n=chris@nfw.ecos.at) |
| 07:10.00 | jplank | what exactly should I see in a sip show channels? |
| 07:10.35 | drmessano^ | channels of the type "SIP" being shown |
| 07:10.41 | drmessano^ | Hence the command anatomy |
| 07:10.46 | jplank | I get that |
| 07:10.52 | jplank | but shouldn't it only be active channels? |
| 07:10.53 | drmessano^ | Youre welcome then |
| 07:10.59 | drmessano^ | yeah |
| 07:11.01 | drmessano^ | Of course |
| 07:11.14 | jplank | what is considered an active channel? |
| 07:11.18 | *** join/#asterisk datacompboy (n=datacomp@l64-89-222.cn.ru) |
| 07:11.18 | drmessano^ | Active SIP channels =! calls |
| 07:11.36 | jplank | I get that |
| 07:12.25 | jplank | but I should see notify's, bye's, subscribes, ect? |
| 07:12.37 | drmessano^ | So why are you even asking? My next comment was going to be "Any SIP communications occuring as part call negotiation, registration, etc", but seems you already know all this |
| 07:13.28 | jplank | maybe I'm misinterpreting "active channels" then |
| 07:13.48 | jplank | why would a bye be considered an "active" channel? |
| 07:14.24 | drmessano^ | If it's communicating, its active |
| 07:14.29 | drmessano^ | active means "doing stuff" |
| 07:14.34 | drmessano^ | as opposed to "not doing stuff" |
| 07:14.59 | jplank | but wouldn't a bye be the end of a communication? |
| 07:15.12 | drmessano^ | SO? |
| 07:15.20 | drmessano^ | It's still a communication |
| 07:15.25 | drmessano^ | Its an ACTION |
| 07:15.27 | drmessano^ | ACTIVE |
| 07:15.38 | drmessano^ | [03:11] <jplank> I get that <-- You sure? |
| 07:15.44 | jplank | it was active when it was sent |
| 07:15.49 | jplank | but now its not |
| 07:15.55 | jplank | thats the point I don't get |
| 07:15.55 | *** join/#asterisk ibm2 (n=Administ@196.203.192.179) |
| 07:17.22 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 07:19.45 | jplank | im an idiot |
| 07:19.51 | *** join/#asterisk mateo_au_ (n=chatzill@c122-106-221-182.belrs3.nsw.optusnet.com.au) |
| 07:20.19 | jplank | I should of looked at the header of the last column |
| 07:32.20 | *** join/#asterisk magenbrot (n=magenbro@ov.odn.de) |
| 07:34.57 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
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| 07:36.49 | *** part/#asterisk blogbasti (n=blogbast@calypso.planet-ic.de) |
| 07:51.07 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 07:51.42 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 07:52.36 | *** join/#asterisk pluesch0r (n=pluesch0@213.185.179.42) |
| 07:54.11 | *** join/#asterisk iwws (n=iwws@213.129.44.243) |
| 07:54.19 | iwws | !ru |
| 07:57.34 | datacompboy | iwws: ? |
| 08:01.36 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 08:04.45 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:19.24 | pluesch0r | what do i need to set to get rid of: "chan_sip.c:3527 process_sdp: Unable to lookup host in c= line, 'IN IP4" ? |
| 08:19.58 | *** part/#asterisk sluxor (n=sluxor@d58-110-243-84.per6.wa.optusnet.com.au) |
| 08:20.18 | pluesch0r | i've set externalip, externalhost and that rvlookup=no .. |
| 08:20.31 | pluesch0r | still, no luck, asterisk is still trying to do a rvlookup. |
| 08:21.27 | pluesch0r | oh. srvlookup .. |
| 08:22.02 | *** join/#asterisk shinao1 (n=shinao1@193.238.30.36) |
| 08:30.14 | *** join/#asterisk sluxor (n=sluxor@d58-110-243-84.per6.wa.optusnet.com.au) |
| 08:30.29 | sluxor | Is there any way I can use asterisk to achieve a video conference between 3 or more people? |
| 08:30.55 | sluxor | what protocols, what codecs, what plugins? |
| 08:31.02 | sluxor | and most importantly what clients support it? |
| 08:33.02 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 08:38.56 | msalaque | sluxor: video codecs H.261, H.263, and H.263+ are supported as far i know |
| 08:39.09 | msalaque | but never tired . |
| 08:39.58 | *** join/#asterisk saftsack (n=saftsack@ip-77-24-133-90.web.vodafone.de) |
| 08:39.58 | sluxor | 3-way video conference? |
| 08:40.03 | sluxor | hmmm... |
| 08:40.27 | sluxor | that would be great because asterisk can integrate with our existing open fire jabber server.. |
| 08:43.35 | sluxor | msalaque what clients would support such feature? suggestion? |
| 08:45.30 | *** join/#asterisk steliosk (n=Stelios@rouss--1.static.otenet.gr) |
| 08:53.52 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 08:54.44 | msalaque | sluxor: sorry was away . i never tried video conferencing in Asterisk . |
| 08:55.10 | msalaque | sorry even i don't have any exp on video . i only work with voice |
| 08:55.17 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-94-244.w86-215.abo.wanadoo.fr) |
| 08:55.53 | msalaque | sluxor: you could try HUD i saw some short of video feature there. |
| 08:55.56 | *** join/#asterisk Vale-ICS (n=vale@icsnet.demon.co.uk) |
| 08:59.14 | *** part/#asterisk saftsack (n=saftsack@ip-77-24-133-90.web.vodafone.de) |
| 09:09.55 | *** join/#asterisk mateo_au (n=chatzill@c122-106-221-182.belrs3.nsw.optusnet.com.au) |
| 09:21.23 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:31.26 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 09:31.50 | *** join/#asterisk steliosk (n=Stelios@91.140.124.241) |
| 09:32.25 | *** join/#asterisk ToTo (n=ToTo@209.8.41.213) |
| 09:36.19 | *** join/#asterisk mateo_au (n=mateo@c122-106-221-182.belrs3.nsw.optusnet.com.au) |
| 09:38.36 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 09:41.59 | *** part/#asterisk sluxor (n=sluxor@d58-110-243-84.per6.wa.optusnet.com.au) |
| 09:49.37 | *** join/#asterisk Blackvel (n=blackvel@dslb-088-065-113-194.pools.arcor-ip.net) |
| 09:52.23 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-251-220.revip2.asianet.co.th) |
| 09:52.59 | *** join/#asterisk SanityIO (n=SanityIO@77.242.105.2) |
| 10:03.56 | miloux | anyone have a good multi function fax/printer/copy machine for SIP to recommend |
| 10:04.35 | *** join/#asterisk spike008t (n=Spoukie@ven69-2-82-228-116-153.fbx.proxad.net) |
| 10:04.43 | petererer | hmm, so, any app_swift users? |
| 10:05.02 | petererer | i can use Allison-8kHz fine, but not Millie(-8kHz) :o |
| 10:05.27 | *** join/#asterisk spike008t (n=Spoukie@ven69-2-82-228-116-153.fbx.proxad.net) |
| 10:06.25 | spike008t | hi all |
| 10:06.44 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 10:06.51 | spike008t | I just want to know if someone has already make work app_mixer? |
| 10:07.27 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 10:08.13 | krdian | miloux: i'm using hp m2727 connected via grandsteram ht386, works fine |
| 10:08.28 | miloux | krdian: thanks |
| 10:09.51 | petererer | hmm |
| 10:28.29 | *** join/#asterisk pcrack (n=pcrack@116.50.213.210) |
| 10:29.04 | *** join/#asterisk vaq (n=xaafwa@vaq/unaffiliated) |
| 10:29.15 | vaq | where should i place my wav file i want to use in my auto dial-out ? |
| 10:29.37 | pcrack | hi..i am configuring a asterisk with vicidial...i have hear a list of leads to be called..but i need customized my lead loader in able to upload the leads |
| 10:29.44 | pcrack | here i mean |
| 10:29.44 | kaldemar | vaq: how does your auto dial-out play files? |
| 10:30.14 | vaq | kaldemar: Channel: Local/1000@from-internal |
| 10:30.14 | vaq | <PROTECTED> |
| 10:30.14 | vaq | <PROTECTED> |
| 10:30.14 | vaq | <PROTECTED> |
| 10:30.14 | vaq | <PROTECTED> |
| 10:30.15 | vaq | <PROTECTED> |
| 10:30.20 | pcrack | can anyone point me to the right direction |
| 10:30.25 | vaq | ups, i want to paste the link http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out |
| 10:34.47 | vaq | Oct 9 12:33:38 WARNING[18476] pbx_spool.c: Unable to open /var/spool/asterisk/outgoing/test.call: Permission denied, deleting |
| 10:34.59 | vaq | the file is chmod 777 and owned by asterisk, why do i still get permission denied? |
| 10:37.06 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 10:37.08 | chrisq | I'm having problems with the "macro queue-addremove" in queues-with-callback-members.txt, the switch seems to just run all alternatives instead of actually being a switch, any ideas? |
| 10:37.22 | vaq | -rwxrwxrwx 1 102 105 107 Oct 9 12:33 test.call |
| 10:41.38 | kaldemar | where does it say the file is owned by asterisk? |
| 10:42.51 | vaq | kaldemar: because 102 is the uid and 105 is the gid of the user asterisk |
| 10:43.09 | kaldemar | ok, if you say so. |
| 10:43.59 | kaldemar | well, app Playback takes a sound file name and looks for the file in /var/lib/asterisk/sounds/ by default. |
| 10:44.29 | *** join/#asterisk mort_gib (n=mjensen@dsl-p4-177.gibconnect.com) |
| 10:44.35 | kaldemar | filename without the extension. "my.wav" would be only "my". |
| 10:45.08 | *** join/#asterisk angryuser (n=Miranda@51.210.20.81.dynamic.adsl.abo.nordnet.fr) |
| 10:45.26 | vaq | Is it possible to make asterisk play a sound when I press dial number@extensions and the person takes the call? |
| 10:45.30 | vaq | i tried it with: http://pastebin.com/m5ac1f56e |
| 10:45.50 | vaq | Shouldn't that play song1 in directory http://pastebin.com/m5ac1f56e twice when the call is answered? |
| 10:46.37 | kaldemar | vaq: yes, it is possible. "core show application dial" will point you in the right direction. |
| 10:47.13 | kaldemar | no, that doesn't play anything. |
| 10:47.28 | vaq | what am i missing? |
| 10:47.53 | kaldemar | the fact that the call never gets there. |
| 10:48.16 | kaldemar | ~thebook |
| 10:48.17 | 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 |
| 10:48.40 | kaldemar | ^ that is a good reference for dialplan logic, you should do some reading. |
| 10:49.20 | vaq | I am missing "Answered" right? because right now i can dial with that extension, the only thing that not works is the sound play |
| 10:49.51 | kaldemar | no. you're missing everything reagarding your dialplan. |
| 10:50.29 | *** join/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 10:50.56 | kaldemar | when a call is answered, the caller is still in the Dial app and doesn't move on in the dialplan until the call is hung up. |
| 10:51.43 | vaq | hmm |
| 10:51.50 | UnixDawg | ? |
| 10:51.56 | vaq | so as it is right now the song is first played after i hang up? |
| 10:52.00 | vaq | (technical) |
| 10:52.09 | kaldemar | it isn't played at all. |
| 10:52.32 | UnixDawg | sorry just getting here |
| 10:52.33 | mchou | can someone provide me with a sanity check that sipbroker responds with busy for all numbers now? |
| 10:53.00 | UnixDawg | s,1,Answer() |
| 10:53.10 | UnixDawg | grrr |
| 10:53.23 | vaq | UnixDawg: ? |
| 10:53.31 | UnixDawg | wrong window |
| 10:53.44 | UnixDawg | ok vaq explain what your doing |
| 10:54.25 | kaldemar | vaq: Dial(SIP/musimi/${EXTEN},120,A(soundfile)) would play an announcement to the called party upon answer. |
| 10:54.38 | kaldemar | he should be studying dialplans. |
| 10:54.48 | *** join/#asterisk gones (n=gones@121.34.20.157) |
| 10:55.17 | vaq | What is the default sound directory? |
| 10:55.25 | kaldemar | i gave it to you already. |
| 10:55.27 | UnixDawg | depends on your setup |
| 10:55.42 | vaq | kaldemar: ye sorry |
| 10:55.42 | UnixDawg | what flavor of linux/bsd |
| 10:55.50 | kaldemar | /var/lib/asterisk/sounds/ by default. |
| 10:55.50 | vaq | Debian |
| 10:55.54 | vaq | but /var/lib/asterisk/sounds/ doesnt exist |
| 10:56.03 | vaq | mkdir /var/lib/asterisk/sounds/; chown asterisk.asterisk /var/lib/asterisk/sounds/ ? |
| 10:56.16 | UnixDawg | look in /usr/share/asterisk |
| 10:56.18 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 10:56.27 | UnixDawg | I believe thats where deb puts it |
| 10:56.57 | chrisq | I'm having problems with the "macro queue-addremove" in queues-with-callback-members.txt, the switch seems to just run all alternatives instead of actually being a switch, any ideas? |
| 10:56.58 | kaldemar | look for astvarlibdir in /etc/asterisk/asterisk.conf, sounds dir should be under it. |
| 10:57.45 | vaq | kaldemar: -- Playing 'vm-goodbye' (language 'en') |
| 10:57.49 | vaq | however i hear nothing |
| 10:58.14 | vaq | exten => _X.,1,Dial(Sip/musimi/${EXTEN},120,A(vm-goodbye)) |
| 10:58.16 | vaq | is my exten |
| 10:59.02 | UnixDawg | thy changing x. to *13 |
| 10:59.26 | vaq | exten => _*13,1,Dial(Sip/musimi/${EXTEN},120,A(vm-goodbye)) |
| 10:59.27 | vaq | <PROTECTED> |
| 10:59.33 | UnixDawg | *12,1,Answer() |
| 10:59.49 | vaq | what? |
| 10:59.52 | UnixDawg | *12,2,PlayBack(filename) |
| 11:00.06 | UnixDawg | *12,3,Hangup() |
| 11:00.17 | UnixDawg | if that works then move forward |
| 11:00.42 | vaq | you mean exten => _ *12,1,Answer() ? |
| 11:00.55 | UnixDawg | Answer tells it to answer the call |
| 11:01.04 | UnixDawg | no _ |
| 11:01.39 | UnixDawg | but yeas add the exten = |
| 11:01.51 | UnixDawg | om 1.4 you no longer need => |
| 11:01.56 | UnixDawg | = works just fine |
| 11:02.55 | vaq | so UnixDawg: http://pastebin.com/m6fc974a4 |
| 11:02.56 | vaq | right? |
| 11:03.28 | UnixDawg | no |
| 11:03.38 | UnixDawg | you padawand |
| 11:03.59 | UnixDawg | your going to far to fast |
| 11:04.19 | vaq | but your extensions would do the same as exten => _*13,1,Dial(Sip/musimi/${EXTEN},120,A(vm-goodbye)) |
| 11:04.28 | vaq | sorry _X. |
| 11:04.54 | UnixDawg | dood |
| 11:05.05 | UnixDawg | have you ever writen dial plan |
| 11:05.18 | UnixDawg | SIP not Sip/ |
| 11:05.53 | UnixDawg | SIP/100 not Sip/1000 |
| 11:05.57 | vaq | UnixDawg: "exten => _X.,1,Dial(Sip/musimi/${EXTEN},120,A(vm-goodbye))" is saying: Playing 'vm-goodbye' (language 'en') when i hit dial number@extension |
| 11:06.10 | vaq | in the asterisk cli |
| 11:06.35 | vaq | doesn't A(vm-goodbye) stands for answer with vm-goodbye ? |
| 11:06.42 | UnixDawg | no |
| 11:06.58 | UnixDawg | not that i recal |
| 11:07.10 | UnixDawg | grabbing the handbook on dial plan |
| 11:08.00 | UnixDawg | _X.= anything diialed |
| 11:08.12 | UnixDawg | start with baby steps |
| 11:08.53 | *** join/#asterisk saftsack (n=saftsack@ip-90-186-229-154.web.vodafone.de) |
| 11:09.45 | UnixDawg | ok you on 1.4 or 1.6 |
| 11:09.48 | UnixDawg | asterisk |
| 11:10.23 | vaq | Asterisk 1.2.13 |
| 11:10.34 | UnixDawg | dood thats way old |
| 11:10.46 | UnixDawg | come into the now and update o 1.4 |
| 11:10.56 | UnixDawg | things have changed |
| 11:11.17 | vaq | can't it be done with 1.2.13 ? |
| 11:11.20 | UnixDawg | thats even old for 1.2 |
| 11:11.48 | UnixDawg | upgrade to a newer version |
| 11:12.15 | vaq | okay |
| 11:12.17 | vaq | one moment |
| 11:12.59 | UnixDawg | you will need toupdate libpri/zaptel/asterisk/asterisk addons |
| 11:13.08 | UnixDawg | and learn to build it correctly |
| 11:13.35 | vaq | I am upgrading the whole asterisk package trough lenny |
| 11:13.56 | UnixDawg | its better to build it yourself to suit your needs |
| 11:14.16 | UnixDawg | using pkgs that someone else built can break your system |
| 11:14.40 | UnixDawg | learn to build a proper system |
| 11:14.58 | vaq | Iknow but this is just a small pbx system that needs to call a number if an alarm is triggered |
| 11:16.03 | vaq | i am upgrading to 1.4.21 |
| 11:16.33 | *** join/#asterisk gazzerh (n=garryh@host-212-158-227-149.dslgb.com) |
| 11:16.39 | vaq | <kaldemar> vaq: Dial(SIP/musimi/${EXTEN},120,A(soundfile)) would play an announcement to the called party upon answer. |
| 11:16.40 | UnixDawg | there are apps out there like pager that will page your cell/pager when a system goes down/alarm goes off |
| 11:16.42 | vaq | so it should work? |
| 11:17.27 | UnixDawg | I dont believe it worked on 1.2.13 . but I have not done any 1.2 coding in a long time |
| 11:17.48 | UnixDawg | brb dog needs out |
| 11:21.43 | vaq | <PROTECTED> |
| 11:21.43 | vaq | Asterisk 1.4.21.2~dfsg-1+b1 |
| 11:22.31 | vaq | UnixDawg: so I did the upgrade, ran dial number@extension which include "exten => _X.,1,Dial(Sip/musimi/${EXTEN},120,A(vm-goodbye))" the console said: -- SIP/musimi-081d1650 answered Console/dsp > -- <SIP/musimi-081d1650> Playing 'vm-goodbye' (language 'en') |
| 11:22.36 | vaq | however i heard nothing |
| 11:23.37 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 11:24.01 | UnixDawg | find out where your sound files are and fix your asterisk.conf |
| 11:24.13 | UnixDawg | look in /usr/share/asterisk/sounds |
| 11:24.28 | UnixDawg | ./val/lib/asterisk/sounds |
| 11:24.34 | *** join/#asterisk LND (n=LND@ppp089210235244.dsl.hol.gr) |
| 11:24.37 | UnixDawg | s/val/var |
| 11:24.43 | vaq | It is /usr/share/asterisk/sounds |
| 11:24.52 | vaq | and it cointains vm-goodbye.gsm |
| 11:25.13 | UnixDawg | then look in your /etc/asterisk/asterisk.conf and make sure its set for that |
| 11:25.34 | vaq | it ain't set anywhere |
| 11:25.44 | vaq | is the prefix astsoundir? |
| 11:25.52 | vaq | astsounddir* |
| 11:27.24 | vaq | it aint set |
| 11:27.45 | UnixDawg | look in /etc/asterisk/asterisk.conf |
| 11:27.51 | vaq | i am in it |
| 11:27.58 | UnixDawg | its not set in there |
| 11:28.03 | vaq | nope |
| 11:28.10 | UnixDawg | thats a issue |
| 11:28.22 | vaq | Iknow, but what is the prefix? is it astsounddir? |
| 11:28.26 | UnixDawg | thats why I said you should build your own |
| 11:28.39 | vaq | http://pastebin.ca/1223638 is my asterisk.conf atm |
| 11:29.45 | UnixDawg | brb |
| 11:30.08 | vaq | what is the prefix for setting the sound directory? |
| 11:30.33 | *** join/#asterisk bminish (n=bminish@2001:770:180:0:0:0:0:10) |
| 11:33.09 | vaq | UnixDawg: should it show an error if it couldn't play the vm-goodbye sound file? |
| 11:33.23 | vaq | UnixDawg: and ain't the default sound directory hard-coded in the configuration somewhere? |
| 11:33.33 | *** join/#asterisk DarkRift (n=dark@65.92.250.231) |
| 11:33.39 | vaq | UnixDawg: I can't find any info at voip-info.org regarding sound directory in asterisk |
| 11:35.47 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 11:37.25 | UnixDawg | ok lets back up here |
| 11:37.59 | UnixDawg | 1 check your trunk settings make sure you have ulaw and gsm codecs enabled |
| 11:38.22 | UnixDawg | if you dont have the gsm codec enabled you wont hear gsm audio files |
| 11:38.45 | UnixDawg | because it wont transcode |
| 11:38.55 | vaq | ah |
| 11:39.00 | *** join/#asterisk Vale-ICS (n=vale@icsnet.demon.co.uk) |
| 11:39.11 | UnixDawg | just hit me while in the shower |
| 11:39.22 | UnixDawg | brb |
| 11:39.29 | vaq | Where should i allow gsm codec and ulaw? |
| 11:40.04 | vaq | iax.conf |
| 11:40.23 | kaldemar | vaq: 13:56 < kaldemar> look for astvarlibdir in /etc/asterisk/asterisk.conf, sounds dir should be under it. |
| 11:41.42 | vaq | says /var/lib/asterisk/sounds so vm-goodbye.gsm should be in that directory |
| 11:41.44 | vaq | am i right? |
| 11:41.45 | *** join/#asterisk setkeh (n=setkeh@CPE-124-180-146-148.vic.bigpond.net.au) |
| 11:41.53 | UnixDawg | listen |
| 11:42.06 | vaq | astvarlibdir => /var/lib/asterisk |
| 11:42.11 | UnixDawg | it this dialing out a sup trunk or just to a extension |
| 11:42.12 | vaq | and i got /var/lib/asterisk/sounds |
| 11:42.41 | UnixDawg | /var/lib needs to be changed on deb to /usr/share/asterisk |
| 11:43.17 | UnixDawg | deb does not install like a normal linux |
| 11:43.46 | UnixDawg | they try to play like bsd does |
| 11:43.59 | vaq | okay: astvarlibdir => /usr/share/asterisk |
| 11:44.02 | vaq | i did restart asterisk |
| 11:44.11 | vaq | and tried dialing again, says: Playing 'vm-goodbye' (language 'en') |
| 11:44.13 | vaq | but it hear none |
| 11:44.22 | UnixDawg | I think /var/lib is a horriable place to put the asterisk stuff |
| 11:44.38 | *** join/#asterisk iwws (n=iwws@213.129.44.243) |
| 11:44.51 | UnixDawg | did you enable the gsm codec on the exten your dialing from |
| 11:44.59 | UnixDawg | and to |
| 11:45.13 | vaq | i did allow=gsm in iax.conf |
| 11:45.24 | subdolus | if I 'cvs get -r OPENBSD_3_9 -P ports |
| 11:45.32 | UnixDawg | ok wait |
| 11:45.39 | UnixDawg | you using aix or sip |
| 11:45.43 | vaq | SIP |
| 11:45.53 | UnixDawg | then you have to put it in the sip.conf |
| 11:46.00 | UnixDawg | not iax.con |
| 11:46.02 | UnixDawg | f |
| 11:46.17 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 11:46.19 | vaq | doh |
| 11:46.22 | vaq | of course |
| 11:46.34 | UnixDawg | you need to go read and learn |
| 11:46.34 | vaq | UnixDawg: allow=gsm under [general] in sip.conf right? |
| 11:46.38 | UnixDawg | no |
| 11:46.49 | UnixDawg | you have to put it on the extension your dialing |
| 11:46.57 | vaq | okay |
| 11:47.02 | UnixDawg | you need to go read the book and dialplan basics |
| 11:47.19 | UnixDawg | have you ever used asterisk before |
| 11:47.22 | UnixDawg | hard core |
| 11:48.02 | UnixDawg | bbiab conf call time |
| 11:48.15 | vaq | UnixDawg: i placed allow=gsm under my ext and did re-dial |
| 11:50.14 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 11:50.52 | vaq | still the same |
| 11:53.25 | vaq | any ideas` |
| 11:57.00 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 11:57.24 | *** join/#asterisk Alocado (n=matthias@dslb-088-068-033-075.pools.arcor-ip.net) |
| 11:57.27 | Alocado | hello |
| 11:57.30 | spike008t | vaq: What is ur problem? |
| 11:57.33 | spike008t | hi |
| 11:57.46 | vaq | spike008t: i am trying to make a call from cli and make it play a sound when answered |
| 11:58.13 | vaq | spike008t: when i answer the call the cli says: <SIP/musimi-081d0e00> Playing 'vm-goodbye' (language 'en') - However i hear nothing in my phone |
| 11:58.26 | vaq | spike008t: ext: exten => _X.,1,Dial(Sip/musimi/${EXTEN},120,A(vm-goodbye)) |
| 11:58.40 | v4mp | hey guys where abouts would i look in book to assign a device without make it the username ? |
| 12:03.14 | Alocado | i have 2 sip accounts registered with my asterisk... if a call is coming in on the one, how can i redirect this call to a predefined number via the other sip account? |
| 12:10.13 | spike008t | vaq: I dunno why it doesn't work... I'll check on my side... |
| 12:10.57 | vaq | hmm i got an error: |
| 12:10.58 | vaq | [Oct 9 13:59:21] WARNING[28724]: chan_oss.c:682 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory |
| 12:11.01 | vaq | does this matter? |
| 12:11.24 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 12:11.59 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 12:11.59 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:12.29 | Blackvel | anyone using phonesuite? it displays wrong incoming MSN number for peer SIP/patton |
| 12:13.36 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 12:15.08 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:18.13 | spike008t | vaq: this error is about your sound device |
| 12:18.53 | vaq | spike008t: i dont have any sound card in my system, is that needed? |
| 12:19.21 | petererer | modules.conf : noload => chan_oss.so |
| 12:19.59 | rwaite | yawn. i'm starting to think that metered service is par the course for sip trunks |
| 12:20.39 | *** join/#asterisk bbryant (n=Brett_Br@adsl-159-32-97.flo.bellsouth.net) |
| 12:21.02 | vaq | petererer: i got load => chan_oss.so in modules.conf |
| 12:21.27 | petererer | change it to noload then, as you don't want it |
| 12:22.15 | vaq | wtf now i cant make dial |
| 12:22.44 | rwaite | do way instain mother? |
| 12:24.10 | vaq | if I use noload = > chan_oss.so the dial command doesn't exist |
| 12:26.36 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 12:27.25 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 12:29.37 | Alocado | i have 2 sip accounts registered with my asterisk... if a call is coming in on the one, how can i redirect this call to a predefined number via the other sip account? |
| 12:33.24 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 12:33.36 | *** join/#asterisk dprado (n=newbie@189.4.114.57) |
| 12:37.24 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.7) |
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| 12:41.50 | Squeeb | Alocado: exten => sip1context,1,Dial(SIP/Sip2) ? |
| 12:41.54 | [TK]D-Fender | Alocado: Dial |
| 12:41.55 | Squeeb | obviously answer it first |
| 12:43.43 | [TK]D-Fender | quite possibly no need to answer |
| 12:43.52 | Squeeb | ooh |
| 12:44.25 | [TK]D-Fender | Call progress follows through if you don't |
| 12:44.32 | Squeeb | :) |
| 12:45.04 | [TK]D-Fender | Depends if you need to ensure that a provider's service doesn't timeout or hit VM, etc. |
| 12:47.49 | *** join/#asterisk propellerhead (n=yogurt2u@host113.190-136-116.telecom.net.ar) |
| 12:50.11 | *** join/#asterisk saftsack (n=saftsack@ip-90-187-235-54.web.vodafone.de) |
| 12:51.57 | *** join/#asterisk admin0 (n=admin@cm37.delta148.maxonline.com.sg) |
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| 12:53.50 | *** part/#asterisk dprado (n=newbie@189.4.114.57) |
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| 13:04.15 | jblack | Word to the wise. If you're gonna have a kid, have a dumb one. The really smart ones are a serious pain in the ass. |
| 13:05.44 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:05.45 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 13:12.34 | v4mp | ok i've looked through book throughly on the things to do with what im trying to do tryed google etc but how im meant to be doing the queues now without agents i can only see how to do it dynamically on cli but surely it can be done in the config too ? |
| 13:14.12 | [TK]D-Fender | v4mp: member => SIP/V4ampire |
| 13:14.36 | v4mp | when i asked if that was right last nite yousaid no :/ |
| 13:14.47 | [TK]D-Fender | v4mp: There. That was it. there was a blatant sample of a fixed DEVICE being a PERMANENT member of a queue |
| 13:15.21 | [TK]D-Fender | v4mp: No, I didn't. Your terminology was mashed all to hell. |
| 13:15.52 | v4mp | i could onlt find sample for Zap/chan and Agent/no. |
| 13:15.56 | *** join/#asterisk gaetronik (n=gaetan@eficaz2.manquehue.net) |
| 13:16.00 | gaetronik | hi there |
| 13:16.15 | gaetronik | i'm trying to do qos |
| 13:16.40 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:16.40 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:16.49 | gaetronik | the tos_audio value of sip.conf apply to agents too? |
| 13:17.15 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 13:17.30 | [TK]D-Fender | gaetronik: What are "agents"? |
| 13:18.45 | gaetronik | in agents.cof |
| 13:18.48 | gaetronik | chan_agent |
| 13:18.54 | gaetronik | this kind of things |
| 13:19.57 | [TK]D-Fender | gaetronik: Well clealy that value is for SIP devices. "chan_agent" is a wrapper and doesn't inherently mean ANYTHING |
| 13:20.01 | gaetronik | bu t with wireshark i saw this |
| 13:20.22 | gaetronik | [TK]D-Fender, it's a thing not clear to me the chan_agent |
| 13:20.39 | gaetronik | it's a wrapper for sip, no? |
| 13:21.27 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 13:25.56 | gaetronik | anyway it works |
| 13:26.28 | [TK]D-Fender | gaetronik: All Chan_agent does is go through the dialplan. this has no impact on anything. |
| 13:26.39 | gaetronik | the next think is to catch why my f*** switch use 3 bit and not 4 |
| 13:26.45 | [TK]D-Fender | gaetronik: If you call out with SIP, all of its rules apply directly |
| 13:27.06 | gaetronik | [TK]D-Fender, i might look silly |
| 13:27.44 | chrisq | problems with "macro queue-addremove" in queues-with-callback-members.txt on ast1.4.19 is resolved by adding MacroExit() after each 'case:' |
| 13:27.46 | gaetronik | i got it |
| 13:28.00 | gaetronik | fucking morning |
| 13:29.05 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 13:30.49 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:32.13 | gaetronik | does asterisk dea with 802.1p |
| 13:32.50 | Corydon76-dig | What is that, TOS? |
| 13:33.36 | gaetronik | COS |
| 13:33.44 | gaetronik | it's at mac level |
| 13:37.27 | *** join/#asterisk albertoandrade (n=alberto@200.195.161.164) |
| 13:38.27 | *** join/#asterisk albertoandrade (n=alberto@200.195.161.164) |
| 13:42.00 | *** join/#asterisk tali81 (i=434e2716@gateway/web/ajax/mibbit.com/x-1bc09453cf336799) |
| 13:42.50 | tali81 | My asterisk is running : -- Executing [s@Debt:45] Dial("SIP/216.235.135.232-081f1a80", "SIP/arettasip/1xxx284xxxx|20|gL[60000.000000]") in new stack |
| 13:43.18 | tali81 | but when the called party hangs up Dial exits witha non-zero, isnt g supposed to make it continue on when the called party hangs up |
| 13:43.39 | *** join/#asterisk moy (n=moy@nat/ibm/x-de51106018f23bac) |
| 13:43.42 | tali81 | the call does connect its just when the called party hangs up , the callee hears a busy |
| 13:43.52 | tali81 | Anyone know what im doing wrong? |
| 13:44.00 | [TK]D-Fender | tali81: Show us the complete call. |
| 13:45.11 | tali81 | do you need the dial plan out the verbose output |
| 13:45.22 | tali81 | out=or |
| 13:46.20 | gaetronik | i saw a patch for cos support but it was not merged |
| 13:46.25 | tali81 | http://pastebin.com/d25d1292 is the verbose out put |
| 13:48.27 | tali81 | line 62 doesnt trigger until the original caller hangs up also. Just FYI |
| 13:50.03 | *** join/#asterisk c4t3l (n=root@74.95.210.124) |
| 13:50.08 | c4t3l | hello world! |
| 13:51.34 | *** join/#asterisk drmessano^ (n=nonya@pdpc/supporter/active/drmessano) |
| 13:55.26 | [TK]D-Fender | tali81: Your timeout looks funny witht he "." in there.... and how long did the call last? you should enable SIP debug or something to confirm who existed |
| 13:55.52 | spike008t | Does Anyone has already made worked application app_mixer? with media mixer? :) |
| 13:55.58 | [TK]D-Fender | tali81: and please include your actual dialplan |
| 13:56.06 | tali81 | ok one second |
| 13:57.35 | tali81 | http://pastebin.com/m421260b2 |
| 13:58.50 | [TK]D-Fender | tali81: Your dialplan does look fine. I would seriously look at the call itself. SIP traffic, etc... |
| 13:59.23 | [TK]D-Fender | tali81: And I've never seen a reported failure of "g" that wasn't due to the call terminating different than expected |
| 13:59.50 | tali81 | Well im dialing asterisk with a land line, and its bridging to my cell. Then i hang up the cell and the land line gets a busy immediatly |
| 13:59.58 | tali81 | with no new data in the verbose log for the cli |
| 14:00.34 | *** join/#asterisk ccesario (n=ccesario@linux.unialco.com.br) |
| 14:00.44 | tali81 | If i then hangup the landline i proceed to the "h" extension |
| 14:01.10 | tali81 | But i was wanting asterisk to force a hangup of the land line when the call disconnected. |
| 14:01.17 | tali81 | So the EU wont hear a busy |
| 14:02.29 | [TK]D-Fender | tali81: Verbose won't give you any more. Go look at the SIP debug |
| 14:02.37 | tali81 | ok |
| 14:04.32 | tali81 | Wierd i was just in sip set debug , and when i hung up the cell, there was still no new entries added by SIP either |
| 14:05.01 | tali81 | like the trunk wasnt notifing asterisk the caller hung up, but something trigger a busy signal somewhere |
| 14:05.20 | tali81 | Do you think i should contact my trunk provider on this then? |
| 14:06.20 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 14:07.02 | Blackvel | when someone calls into the IVR |
| 14:07.24 | Blackvel | is there any way to bring him out of IVR and pickup him up at snom phone? |
| 14:08.38 | [TK]D-Fender | tali81: Sounds line an inband busy, and if that's what is happening its your outbound provider's fault and unless the fix it, you're screwed |
| 14:10.15 | tali81 | ok ill contact them then. Thanks for the help, i was really stumped on this |
| 14:10.22 | [TK]D-Fender | tali81: np. |
| 14:14.05 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 14:14.17 | Zeeek | i |
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| 14:30.20 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 14:31.40 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:31.57 | ChrisN | I'm using a custom voicemail application and trying to get Message Wait Indicator (MWI) working. Is there a command that will send a packet for this, or a way to send custom SIP packets to agents using SIP with nat=yes? |
| 14:32.33 | jaytee | custom voicemail? you mean external voicemail system? |
| 14:33.05 | ChrisN | I'm using an AGI rather than using Asterisk's default voicemail application. |
| 14:33.38 | jaytee | oh, ok. |
| 14:34.17 | *** join/#asterisk VijayG (n=vijay@58.68.47.109) |
| 14:34.33 | jaytee | I'm using Exchange 2007 UM so I've been looking for a way to get MWI on my phones myself but UM doesn't send the SIP notify packet. |
| 14:34.56 | *** part/#asterisk VijayG (n=vijay@58.68.47.109) |
| 14:34.57 | ChrisN | I was going to use sipsak to send the MWI sip packets, but some users are behind NAT. |
| 14:35.33 | jaytee | that's what I was looking at doing but it's not a workable solution for all cases in my environment. |
| 14:35.59 | jaytee | or at least not 100% reliable |
| 14:36.28 | shriven | Hello, I believe i have a simple question that I have not been able to find an answer to... Is it possible to use some form of pattern matching in my extensions to match named extensions? Ie: user sean has extension 6000, user john has extension 6001, I can match the two numbers with _XXXX, is it possible to match alpha characters with a similar method? Ie: using a regex in the extension? |
| 14:36.36 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 14:36.51 | jaytee | we have multiple users in some areas sharing the same computers and accounts. I've been toying with the idea of using rules in Outlook to run a vb script that kicks off a SIPSAK MWI when ever newmail comes into Outlook with a voicemail attachment. |
| 14:36.54 | [TK]D-Fender | shriven: No |
| 14:36.58 | shriven | blast |
| 14:37.24 | shriven | so if I want to use named extension dialing, I am stuck with literally writing everybody's name into the dialplan? |
| 14:38.14 | [TK]D-Fender | shriven: The pattern match system sucks. The second you use "_" to start it, "x" & "X" get reserved and so much more. |
| 14:38.32 | russellb | you can match alpha patters, but in the same way you do numbers ... like .... _[abcdefg][abcdefg][abcdefg] |
| 14:38.33 | russellb | :-p |
| 14:38.34 | [TK]D-Fender | shriven: This FUBAR's a lot of ideas before they get off the ground |
| 14:39.00 | [TK]D-Fender | russellb: We need to ditch our pattern syntax for regex... seriously./ |
| 14:39.05 | shriven | right, I was hoping there was some other method I was just missing. like somehow using a regex |
| 14:39.07 | russellb | yeah yeah yeah |
| 14:39.13 | shriven | that sucks a lot. : ( |
| 14:39.15 | russellb | punt to AGI |
| 14:39.19 | shriven | ok thanks for the answer |
| 14:39.22 | [TK]D-Fender | russellb: then there is the application parameter parser..... |
| 14:39.36 | russellb | i'm not here looking for advice :-p |
| 14:39.38 | shriven | ressellb: alpha patters? |
| 14:39.47 | [TK]D-Fender | russellb: punting to AGI doesn't solve it because you'd need a catch-all that will guarantee a certain amount of progress... |
| 14:40.00 | shriven | you mean, put a [a-z] sort of thing for every possible username character? |
| 14:40.06 | russellb | yeah, what i said would match a 3 letter extension (only letters a through g because i didn't feel like typing more) |
| 14:40.18 | shriven | ugh |
| 14:40.24 | russellb | just saying what's there. |
| 14:40.27 | shriven | yeah, thanks |
| 14:40.28 | [TK]D-Fender | russellb: Depending on your needs, it might do, but then again, * standard extens would CLASH. Think about "h"! z0mg, nightmare waiting to happen |
| 14:40.30 | shriven | that is good to know |
| 14:40.42 | shriven | right |
| 14:41.03 | jaytee | shriven, just out of curiosity, what are you going to dial FROM, not a standard IP phone I hope? |
| 14:41.05 | shriven | it would be much better if I could do some sort of match on any amount of [:alpha:] |
| 14:41.16 | shriven | jaytee: softphones |
| 14:41.22 | jaytee | ah, ok |
| 14:41.40 | shriven | it's not a big deal if I can't do it... but as 90% of my company is going to use softphones when this is all running... it would be nice |
| 14:42.58 | jaytee | shriven, there's always MS OCS 2007 and Office Communicator. :-) |
| 14:45.23 | shriven | ressellb, [TK]D-Fender: related to using [abcd] etc, will that match a pattern that has fewer than the specified [abcdefghi] characters specified? So if I had extension "abc" and I specified _[abc][abc][abc][abc] would I get a hit? |
| 14:45.32 | shriven | jaytee: all apple here. |
| 14:48.18 | shriven | russelb: sorry mis typed your name there... |
| 14:49.45 | *** join/#asterisk hfb (n=hfb@pool-96-247-117-68.lsanca.dsl-w.verizon.net) |
| 14:50.20 | *** join/#asterisk seanmh (i=HydraIRC@216.31.101.93) |
| 14:52.00 | Squeeb | shriven: Mac Mini PBX ? :) |
| 14:52.05 | Squeeb | That's what we have here |
| 14:52.07 | Squeeb | perfect for it |
| 14:52.23 | shriven | squeeb: actually it's on an older dual cpu G4 |
| 14:52.29 | Squeeb | ah |
| 14:52.34 | Squeeb | ours is on a single G4 .. |
| 14:52.38 | Squeeb | seems pretty happy with it though |
| 14:52.54 | shriven | good to know. : ) I haven't really gotten to testing it with heavy load yet |
| 14:53.07 | Squeeb | we only handle about 10 calls simultaniously max |
| 14:53.15 | Squeeb | and it never goes above 1.00 load |
| 14:53.45 | *** join/#asterisk ManxPower (n=manxpowe@218.sub-75-202-234.myvzw.com) |
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| 14:56.10 | shriven | excellent, we would probably be in a similar situation |
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| 15:14.59 | jaytee | <PROTECTED> |
| 15:16.11 | shriven | no, I installed debian. |
| 15:16.42 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) |
| 15:20.19 | zeljkoMON | can some1 recomend me nice soft echo cacelation for misdn interface? |
| 15:20.47 | Blackvel | octasic |
| 15:20.53 | Blackvel | olsec |
| 15:21.05 | ManxPower | I didn't know the oslec worked with mISDN |
| 15:21.22 | Qwell | I didn't think mISDN could so swec |
| 15:21.22 | Blackvel | and was not digium hpec also working |
| 15:21.24 | Qwell | do* |
| 15:21.32 | ManxPower | Blackvel: where would you purchase the Octastic one? |
| 15:21.45 | Qwell | ManxPower: my guess is from Octasic :p |
| 15:21.57 | Qwell | and, you make the same typo I do every time... why is that so hard to type? |
| 15:22.01 | Blackvel | I purchased mine (for bristuff) from UK |
| 15:22.01 | zeljkoMON | can u buy it online and whats the price if some1 knows |
| 15:22.03 | ManxPower | I wonder how much it costs? $25,000? |
| 15:22.12 | Blackvel | 10$ / channel |
| 15:22.14 | Blackvel | or was it 20$ |
| 15:22.23 | Blackvel | anyways...couldn't remove my echo problems |
| 15:22.24 | coppice | sometimes EC works with mISDN, but mISDN is so quirky results are erratic |
| 15:22.27 | ManxPower | Blackvel: that is called the HPEC around here. |
| 15:22.38 | Blackvel | exactly...I mean that! |
| 15:22.41 | Blackvel | :) |
| 15:22.43 | Qwell | ManxPower: HPEC != Octasic |
| 15:22.58 | Blackvel | true |
| 15:23.20 | Blackvel | zeljkoMON: was not testing octasic with misdn, but I know beronet offers it |
| 15:23.34 | Blackvel | dunno if it was my epia via nemiah 1 gig cpu |
| 15:23.56 | ManxPower | Qwell: who makes the HPEC? |
| 15:24.12 | Blackvel | anyways...patton gateway works well |
| 15:24.15 | Qwell | us? |
| 15:24.20 | zeljkoMON | i got some intel quad core so it should eork fie |
| 15:24.22 | Qwell | ManxPower: I don't know |
| 15:24.24 | ManxPower | ALL the echo cancelers that I've heard for use with Asterisk only work with Zaptel. |
| 15:24.36 | Blackvel | same for octasic |
| 15:24.49 | *** join/#asterisk jploh (n=jploh@122.53.210.115) |
| 15:24.51 | Blackvel | ehm...it works with zaptel too I mean |
| 15:25.12 | ManxPower | Blackvel: zeljkoMON is using mISDN, not Zaptel |
| 15:25.54 | Blackvel | it works with that |
| 15:26.02 | Blackvel | let me find the url |
| 15:26.07 | coppice | oslec works with mISDN, but mISDN keeps doing funky things with its timing and breaks the canceller |
| 15:26.28 | Qwell | moral of the story? don't use BRI |
| 15:26.45 | Qwell | (yes, I'm kidding) |
| 15:26.55 | coppice | nothing wrong with BRI. something's horribly wrong with mISDN, though |
| 15:26.59 | Blackvel | https://shop.beronet.com/product_info.php/products_id/209 |
| 15:27.04 | ManxPower | Qwell: If they are using Asterisk I think they should avoid BRI. |
| 15:27.43 | Blackvel | Qwell: moral of the story? don't use BRI with misdn/bristuff but get external bri-sip gateway (like patton)! |
| 15:27.49 | Blackvel | that works quite well without echo problems |
| 15:28.26 | Blackvel | I always hear other ppl telling me oslec fixed bristuff echo problems or even bristuff (e.g junghanns duobri/quadbri) has no echos (because of BRI and not old pstn) |
| 15:28.32 | Blackvel | but I tested it myself |
| 15:28.34 | Blackvel | and throw it away |
| 15:29.02 | Blackvel | dunno if it is only marketing at well. it may depend on so many factors |
| 15:29.27 | Blackvel | I can get the patton gateways at nearly the same price as this non HW-EC isdn cards |
| 15:29.46 | Blackvel | so why bother and patch/compile bristuff/SW-EC, misdn at all? |
| 15:30.27 | zeljkoMON | the prob is i have to work with hardwer i got :( |
| 15:30.29 | Blackvel | zeljkoMON: so you don't even use MG2 right now? |
| 15:30.40 | Blackvel | just because MG2 is in zaptel |
| 15:30.47 | Blackvel | oh I see |
| 15:30.55 | Blackvel | how many channels has your cards? 4?8? |
| 15:31.04 | zeljkoMON | 4 bris |
| 15:31.20 | Blackvel | I would give octasic a run (beronet runs their isdn cards on misdn) |
| 15:32.53 | ManxPower | Blackvel: echo comes from the FAR end loop. |
| 15:33.04 | ManxPower | so it matters not what your local interface to the PSTN is. |
| 15:33.17 | Blackvel | its the other way around |
| 15:33.23 | Blackvel | I had no problems on snom |
| 15:33.27 | ManxPower | Blackvel: you are wrong. |
| 15:33.40 | Blackvel | but calling my analog pstn or mobile phone got me horrible echo / sound quality on the PSTN side |
| 15:34.05 | ManxPower | echo can be caused by a bad speakerphone or other crappy telephone hardware, but that cannot be removed by any EC. |
| 15:34.16 | Blackvel | thats snom 370 |
| 15:34.22 | Blackvel | no crappy hardware :( |
| 15:34.30 | ManxPower | Blackvel: whatever. |
| 15:34.40 | Blackvel | zeljkoMON: do you hear echo or does the pstn side have it? |
| 15:34.58 | [TK]D-Fender | Blackvel: bad AEC in the firmware can do it. This varied on many GS's for a long time |
| 15:35.31 | mort_gib | Blackvel: I use a lot of Sangoma A500 BRI cards, |
| 15:35.40 | Blackvel | with HWEC? |
| 15:35.45 | mort_gib | Yes |
| 15:35.48 | mort_gib | Great stuff |
| 15:36.06 | Blackvel | finally that works too :) (there was a long time bug in firmware) |
| 15:36.18 | Blackvel | I researched to buy one some weeks ago, too |
| 15:36.32 | Blackvel | was august or something |
| 15:36.34 | mort_gib | No Zaptel required, if you only run Bri, but if you need the odd analogue device |
| 15:36.59 | mort_gib | I installed the first one May, and yes there were issues, not serious ones but they were handled |
| 15:37.18 | Blackvel | [TK]D-Fender: you are so right! a company told me that too. And I really had fear for having to throw away both snom's. but fortunately they where ok with latest firmware |
| 15:37.43 | Blackvel | honestly guys |
| 15:37.50 | mort_gib | Snoms are cool |
| 15:38.07 | Blackvel | the hardest part for my new voip equipment was: what damn card + drivers should I go for. I researched 1-2 weeks |
| 15:38.08 | zeljkoMON | Blackvel i hear echo on voip side |
| 15:38.15 | Blackvel | and then even bought the wrong hardware |
| 15:38.59 | Blackvel | zeljkoMON: oh oh. I am not really convinced that SW-EC will fix that, as [TK]D-Fender said |
| 15:39.00 | mort_gib | Blackvel: Sorry but what hardware did you get? |
| 15:39.08 | Blackvel | patton isdn bri gateway |
| 15:39.11 | *** join/#asterisk skr1p7 (n=r00t@line189-55.adsl.kirov.ru) |
| 15:39.34 | Blackvel | external hardware which connects asterisk as SIP |
| 15:39.54 | mort_gib | Some guys in here suggested Media Gateways... I end up with the Sangoma cards based on recommendation from my hardware pusher |
| 15:40.28 | Blackvel | jupp, patton is media gw |
| 15:40.30 | ManxPower | mort_gib: Apparently some people here think SIP Gateways are sexy. They are not really my type. |
| 15:40.47 | mort_gib | I don't know, I haven't tried them |
| 15:41.14 | mort_gib | I had issues with the Sangoma cards, but I suspect learning curve was involved ;-) |
| 15:41.26 | Blackvel | mort_gib: I heard ppl telling that only A500D (with HWEC) works really. some others tell A500 (without HWEC) works with no problems |
| 15:41.30 | Blackvel | thats the hard party |
| 15:41.37 | Blackvel | really everyone is telling something different |
| 15:41.43 | Blackvel | it MAY or MAY NOT work |
| 15:41.58 | Blackvel | depends on situations, lines, server equipement...whatever |
| 15:42.05 | mort_gib | I also, now, strongly believe that your troubleshooting options are FAR better with hardware installed in the box |
| 15:42.16 | mort_gib | I never tried without HWEC, no point! |
| 15:42.31 | Blackvel | mort_gib: well I ALMOST bought the A500D too :) but it has too many lines (double BRI lines per card) |
| 15:42.32 | Blackvel | hehe |
| 15:42.57 | mort_gib | Blackvel: Well compare the price to a media gw |
| 15:43.01 | Blackvel | now I pissed ManxPower off :) |
| 15:43.05 | Blackvel | cheaper |
| 15:43.06 | *** join/#asterisk ManxPower (n=manxpowe@218.sub-75-202-234.myvzw.com) |
| 15:43.11 | Blackvel | 370 euro netto |
| 15:43.12 | skr1p7 | hi all. I want connect LAN asterisk server to SIPNET. I setup my sip.conf and add extension to dial by SIPNET. But in logs: WARNING[1499]: chan_sip.c:1787 __sip_xmit: sip_xmit of 0x81ca7b0 (len 774) to 212.53.40.40:0 returned -1: Invalid argument |
| 15:43.15 | mort_gib | In any event the HWEC cost more that the Bri modules |
| 15:43.25 | Blackvel | true |
| 15:43.32 | jploh | i have a question with mixmonitor |
| 15:43.39 | Blackvel | zeljkoMON: what voip equipment? |
| 15:43.41 | Blackvel | snoms? |
| 15:43.43 | mort_gib | And... Never go cheap on * hardware |
| 15:43.58 | Blackvel | well 650 to 370 is a difference |
| 15:43.59 | skr1p7 | how to fix that? I'm using NAT. (what ports asterisk need to call out)? |
| 15:44.13 | zeljkoMON | openvox te400p or smth like that |
| 15:44.17 | jploh | if a call is being monitored then the call is transferred, will the monitoring stop? |
| 15:44.23 | zeljkoMON | ans grandstream phones |
| 15:44.48 | Blackvel | zeljkoMON: I read that some ppl had issues with grandstream |
| 15:44.52 | ManxPower | ~sipnat |
| 15:44.53 | 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 |
| 15:44.53 | Blackvel | (echos on phone side) |
| 15:45.04 | Blackvel | [TK]D-Fender seems to have the same opinion |
| 15:45.05 | ManxPower | skr1p7: go read the first link and follow the directions |
| 15:45.27 | Blackvel | zeljkoMON: what side do you call ? analog pstn? mobile? |
| 15:45.41 | Blackvel | zeljkoMON: do you have the echos EVERYTIME? |
| 15:46.20 | Blackvel | zeljkoMON: I just think like [TK]D-Fender that SW-EC won't /may not fix it |
| 15:46.23 | ManxPower | You don't usually have echo on calls to mobile phones or long distance calls. |
| 15:47.07 | ManxPower | (all mobile networks use echo canceling, calls that go thru the long distance network echo canceled. |
| 15:47.30 | zeljkoMON | its like this |
| 15:47.34 | Blackvel | interesting |
| 15:47.37 | zeljkoMON | voip phone to voip phone is ok |
| 15:47.55 | zeljkoMON | but when i dial pstn it has echo |
| 15:48.09 | Blackvel | I can not even say what the most problems where on mobile (with duobri / bristuff)...mabye to weak cpu |
| 15:48.13 | ManxPower | zeljkoMON: there is no analog loop on VoIP calls so you cannot get echo on those calls. |
| 15:48.16 | Blackvel | pstn isdn? analog isdn? dect phone? |
| 15:48.35 | ManxPower | zeljkoMON: do you get echo when you dial a PSTN cell phone? |
| 15:48.45 | zeljkoMON | yes |
| 15:49.04 | ManxPower | zeljkoMON: that is very very very strange since when you call a mobile there is no analog loop involved |
| 15:49.05 | Blackvel | I can not even say WHY the most problems where on mobile (with duobri / bristuff)...mabye to weak cpu |
| 15:49.33 | Blackvel | (only on mobile side, not voip side) |
| 15:49.52 | Blackvel | zeljkoMON: can you get latest firmware? try another grandstream phone? replace one by snom? |
| 15:49.59 | ManxPower | Blackvel: I'm not talking about the person on the other end, I'm talking about the voIP person |
| 15:50.02 | Blackvel | just try to see if its GS side |
| 15:50.21 | Blackvel | ManxPower is so right |
| 15:50.53 | Blackvel | had no echo on voip side when calling mobile (and saying something on mobile side) |
| 15:51.01 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 15:51.13 | [T]ank | anyone in the market for a TE420P? |
| 15:51.31 | ManxPower | [T]ank: I'll give you %40 for it |
| 15:51.38 | ManxPower | I mean $50 |
| 15:51.55 | Blackvel | do you guys really experience it from time to time that snoms/grandstream phone are just defect? and replacing it fixes all the problems? |
| 15:52.10 | [T]ank | lol... you offer me that every time I have something available :-D |
| 15:52.13 | Blackvel | I got that told like 10-20% are always corrupt |
| 15:52.32 | Blackvel | its like you buy 100-200 and not all are okay |
| 15:53.30 | zeljkoMON | can i try with soft-phone? |
| 15:53.32 | [TK]D-Fender | ~gs |
| 15:53.33 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 15:53.55 | Zeeek | is prejudice really extreme in this case? |
| 15:54.11 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 15:54.15 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 15:54.41 | mort_gib | GrandSuck :) |
| 15:55.22 | mort_gib | TK just on that note, my users mostly prefer Snoms... As in the LIKE to play around with the webinterface |
| 16:01.35 | jaytee | I had the hardest time getting an RMA for Grandstream phones because the vendor couldn't understand me due to all the echo and jitter. Once I got my new Polycom 330 setup I was able to finally get an RMA :-) |
| 16:02.15 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:02.57 | *** join/#asterisk zeljkoMON (n=bum@mail.shollex.com) |
| 16:05.45 | Zeeek | "you're breaking up, Sir" |
| 16:07.50 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 16:10.53 | *** part/#asterisk jploh (n=jploh@122.53.210.115) |
| 16:12.41 | jeev | anyone know a good place to get solution manuals ? |
| 16:15.25 | cesar_CR | jeev, about what ? |
| 16:15.44 | hardwire | jeev: ? |
| 16:16.20 | zeljkoMON | situation update voip phone to mobile no echo |
| 16:16.27 | zeljkoMON | to land lines echo |
| 16:16.35 | jeev | a book |
| 16:16.52 | hardwire | solutions of a book, in manual form? |
| 16:17.13 | jeev | yes |
| 16:17.28 | hardwire | make sense. |
| 16:17.30 | hardwire | I command you. |
| 16:18.05 | jeev | Introduction to Environmental Engineering and Science (3rd Ed., Gilbert M. Masters & Wendell P. Ela) |
| 16:19.34 | hardwire | I want a telephone pstn starters guide |
| 16:19.40 | hardwire | so I know how CIC's work and are routed too |
| 16:20.53 | jeev | heh |
| 16:21.36 | *** join/#asterisk zeljkoMON (n=bum@cable-89-216-173-176.dynamic.sbb.rs) |
| 16:21.37 | jeev | the office i have an office at |
| 16:21.39 | jeev | got me a cake |
| 16:21.41 | jeev | goodbye cake lol |
| 16:21.54 | Qwell | cake? why, you quitting? |
| 16:21.58 | jeev | no |
| 16:21.59 | jeev | quitting |
| 16:22.02 | jeev | it's my business |
| 16:22.08 | jeev | i use a desk at their office when i dont want to go to my office |
| 16:22.12 | jeev | i'm going to london for a week! |
| 16:22.20 | hardwire | http://www.logicsupply.com/products/wdt_01 |
| 16:22.22 | hardwire | I need a few of these |
| 16:22.38 | *** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 16:22.38 | *** join/#asterisk CGMChris (n=chris@mail.cgmyes.com) |
| 16:22.54 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 16:22.55 | Qwell | hardwire: neat, how does it know something died? |
| 16:23.02 | hardwire | you're getting a cake for leaving for a week? |
| 16:23.18 | hardwire | Qwell: there are kernel modules that help with that |
| 16:23.24 | Qwell | ahh |
| 16:23.26 | Qwell | serial? |
| 16:23.28 | hardwire | serial watchdog modules |
| 16:23.29 | hardwire | yeh |
| 16:23.39 | hardwire | and the watchdog package can do it as well |
| 16:24.09 | hardwire | have you seen the rhino PRI failover/passive tap ? |
| 16:24.13 | CGMChris | Does anyone know of an argument or a way to call VoiceMail without it playing the unavail or busy message? |
| 16:24.14 | hardwire | I think that's neat too |
| 16:24.25 | hardwire | CGMChris: yes. |
| 16:24.32 | hardwire | CGMChris: but I'm a huge smarty pants. |
| 16:24.41 | CGMChris | hardwire: Please, by all means, do tell. |
| 16:24.41 | Qwell | CGMChris: application show voicemail |
| 16:24.51 | Qwell | or, whatever the command is |
| 16:24.55 | Qwell | core show application voicemail? |
| 16:24.57 | hardwire | CGMChris: doo iiiiit |
| 16:25.21 | CGMChris | oh wow, i didnt even know "core show application" was there. Thanks! |
| 16:25.30 | Qwell | CGMChris: type 'help' |
| 16:25.41 | CGMChris | I got it, thanks! |
| 16:26.07 | hardwire | CGMChris: I hear a little rainbow star noise followed by "The More You Know(tm)" |
| 16:26.17 | Qwell | CGMChris: That'll be $99.95 |
| 16:26.29 | hardwire | Qwell: I was here first |
| 16:26.30 | hardwire | back off |
| 16:26.31 | Qwell | also... |
| 16:26.32 | Qwell | ~book |
| 16:26.34 | jbot | rumour has it, book is 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 or see ~buybook |
| 16:26.42 | Qwell | hardwire: you owe me a million dollars anyways |
| 16:26.47 | hardwire | bah.. who wants lmadsens book.. srsly. |
| 16:26.50 | hardwire | Qwell: you never told me |
| 16:26.58 | hardwire | I owe you a fart in the face |
| 16:26.59 | CGMChris | <- read book, mustve missed the core application show section |
| 16:27.02 | Qwell | did too! you weren't paying attention |
| 16:27.18 | hardwire | Qwell: lies |
| 16:27.21 | hardwire | you don't even know. |
| 16:27.27 | hardwire | it's sooo secret it's a secret. |
| 16:28.15 | ManxPower | CGMChris: It's better to read the docs before the book |
| 16:28.28 | *** join/#asterisk l2trace99 (n=jr@75.112.133.235) |
| 16:29.11 | CGMChris | I just read about that guy that did FreeSwitch... how he did a bunch of work on Asterisk, was given credit in the book, and was then subsequently removed from the book credits after starting a fork. his pbx sys sounds good, but I'm not sure there is a large enough community to support it. Thoughts anyone? |
| 16:29.31 | CGMChris | er, not a fork, but a rewrite. |
| 16:29.57 | hardwire | I sat next to him, I guess, a few times @ astricon |
| 16:30.03 | jeev | yea hardwire, i'm getting a cake for leaving for a week lol |
| 16:30.04 | hardwire | he's nosy |
| 16:30.09 | hardwire | in a good way |
| 16:30.14 | hardwire | jeev: you suck |
| 16:30.23 | hardwire | I'm gone for weeks at a time and I don't even get a hello when I come back. |
| 16:30.28 | CGMChris | I'd rather have a nosy programmer than an indifferent one. |
| 16:30.49 | jeev | haha |
| 16:32.12 | *** join/#asterisk bouil (n=bouillon@yut57-1-88-169-9-192.fbx.proxad.net) |
| 16:32.15 | bouil | hi |
| 16:34.43 | *** join/#asterisk metfan2007 (n=jc@189.135.127.11) |
| 16:36.19 | metfan2007 | hi all! I just installed 50 Aastra SIP phones, but I'm seeing a lot of " Peer '4826' is now UNREACHABLE! Last qualify: 253" for all the phones, the network cables and switch are Ok, how can I avoid this problem? |
| 16:36.34 | ManxPower | metfan2007: remove the qualify= option |
| 16:38.28 | [TK]D-Fender | CGMChris: Not a fork, nor a rewrite, but a completely separate project. |
| 16:40.31 | hardwire | [TK]D-Fender: I'm down for a new project. |
| 16:40.39 | hardwire | I'd love a completely non-portable linux kernel based sip router :0 |
| 16:40.44 | hardwire | how evil! |
| 16:40.45 | hardwire | delightful! |
| 16:40.56 | hardwire | zap to sip bridging in kernel space makes me cry. |
| 16:41.11 | *** join/#asterisk vonkleist (n=vonkleis@189.155.116.80) |
| 16:42.17 | metfan2007 | ManxPower: so what is the qualify option for? |
| 16:42.57 | ManxPower | metfan2007: it watches to see if the remote side goes offline. |
| 16:43.38 | metfan2007 | ManxPower: Ok |
| 16:44.11 | metfan2007 | ManxPower: so it is only recommended y remote phones cases? |
| 16:44.54 | ManxPower | metfan2007: I don't recommend it at all, actually. |
| 16:45.12 | ManxPower | qualify= can be used to deal with NAT UDP timeout, but there are other ways to do tha |
| 16:45.13 | ManxPower | t |
| 16:46.57 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 16:48.34 | angryuser | metfan2007 : better enable it for remote phones, besides keeping port open you can identify some network problems (ping too high, firewall problems &else) |
| 16:52.29 | ManxPower | metfan2007: *I* recommend that you just set your registration interval for 1m or use whatever NAT keepalive options are on the device. |
| 16:52.51 | jameswf | is amaised the spec file for new rpm's based on dahdi are 1/4th the size of the zaptel.... think i need to rewrite the other spec file |
| 16:54.38 | *** join/#asterisk iwws (n=iwws@213.129.44.243) |
| 16:54.49 | metfan2007 | ManxPower: no matter if all the phones are in the same LAN or just for remote phones? |
| 16:57.10 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 17:03.50 | ManxPower | metfan2007: you should not need to do anything for phones on the same LAN |
| 17:05.08 | *** join/#asterisk shinao1 (n=shinao1@193.238.30.36) |
| 17:08.22 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:09.10 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 17:28.21 | *** join/#asterisk viraptor (n=viraptor@awh178.internetdsl.tpnet.pl) |
| 17:28.24 | *** join/#asterisk Firass-VC22 (n=firass@restek-ws-0.vikcomm.wwu.edu) |
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| 17:34.22 | *** part/#asterisk viraptor (n=viraptor@awh178.internetdsl.tpnet.pl) |
| 17:34.50 | *** join/#asterisk stoffell (n=stoffell@d51A4D6CC.access.telenet.be) |
| 17:35.23 | *** join/#asterisk UnixDawg_ (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 17:38.53 | *** join/#asterisk viraptor (n=viraptor@awh178.internetdsl.tpnet.pl) |
| 17:40.30 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 17:40.35 | metfan2007 | ManxPower: I just removed the qualify line in sip.conf for all the phones, but I still see that the phones desregister and register again every certain time, any idea? |
| 17:41.17 | viraptor | a bit OT, but: does anyone know of a page distributing royality free short sounds I could use in my pbx? I'm looking for a 'ping' sound (not the voicemail beep) |
| 17:42.57 | ManxPower | viraptor: "royalty free sounds" in google? |
| 17:43.18 | adr3nalin3 | Could someone please take a look at my dialplan and see if I can do what I am trying to do? I am trying to forward everything that comes in my span 1 to external number. Sorry for the large growth of smegma in my dialplan. http://pastebin.com/m112a4786 |
| 17:43.40 | viraptor | ManxPower: it's mostly spam unfortunately :( I'm already going through 8th or so page of that |
| 17:43.53 | adr3nalin3 | The line I am questioning is: exten = _1333,1,Macro(trunkdial-failover-0.3,${span_1}/93064109,,span_1,span_1) |
| 17:44.05 | jaytee | is very hesitant to look at any dialplan with "smegma" in it |
| 17:44.16 | thehar | haha |
| 17:44.24 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 17:44.29 | jaytee | rowr! |
| 17:44.29 | *** join/#asterisk UnixDawg (n=UnixDawg@155.129.204.68.cfl.res.rr.com) |
| 17:44.36 | Katty | herro. |
| 17:44.41 | jaytee | hi Katty |
| 17:44.44 | adr3nalin3 | herro rara |
| 17:45.34 | ManxPower | viraptor: how about "royalty free music"? I once looked into that for MoH. Maybe the music sites have other sounds. |
| 17:45.54 | viraptor | ManxPower: I'll keep looking, thanks |
| 17:46.20 | jaytee | even though it's intended to mimic kitty speak whenever I see words where R replaces L I think of the Gorgeous Tiny Chicken Machine Show |
| 17:47.26 | Katty | [TK]D-Fender: :> |
| 17:47.35 | Katty | [TK]D-Fender: http://www.new.facebook.com/photo.php?pid=34117767&l=44e95&id=37617946 |
| 17:50.10 | Katty | [TK]D-Fender: http://www.new.facebook.com/photo.php?pid=34117807&id=37617946 <- 9 weeks. |
| 17:51.30 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 17:51.55 | *** part/#asterisk dprado (n=newbie@189.4.114.57) |
| 17:57.46 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 18:00.54 | tzafrir_laptop | ManxPower, the creative commons by-sa license mentions specifically public performance |
| 18:01.38 | tzafrir_laptop | There are a number of voice prompts sets under a cc-by or cc-by-sa license |
| 18:02.17 | *** join/#asterisk unpaidbill (i=bill@420nugs.info) |
| 18:10.16 | *** join/#asterisk zionvier (n=pete@c-67-166-22-106.hsd1.co.comcast.net) |
| 18:11.11 | skr1p7 | how to fix that: NOTICE[21195]: chan_sip.c:12322 handle_response_invite: Failed to authenticate on INVITE to '"SipPhone" <sip:alphanet@sipnet.ru>;tag=as72cd2fde'? at provider's web interface my connection in list of connected devices |
| 18:13.03 | zionvier | Can anyone suggest a good SIP provider for home use in the states? I've seen the list of sip providers on sipcenter.com, but a recommendation would be appreciated. |
| 18:15.37 | StephenF | zionvier: checkout voicepulse |
| 18:15.49 | StephenF | also |
| 18:15.54 | StephenF | ~itsp |
| 18:15.55 | 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 |
| 18:16.02 | StephenF | ~itsplist-us |
| 18:16.03 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 18:16.14 | StephenF | zionvier: ^^ |
| 18:16.26 | *** part/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 18:16.32 | zionvier | thank you |
| 18:16.47 | bouil | ~itsplist-lu |
| 18:16.57 | bouil | :-/ |
| 18:17.04 | bouil | i'm beginning with asterisk, and i want to make a callbak only if the caller number is mine. Do you know how to do that ? for now it works with any caller. i have this line "exten => s,2,Goto(init-callback,s,1)" |
| 18:17.13 | petererer | has anyone found enum services useful? |
| 18:18.37 | [TK]D-Fender | bouil: "core show application gotoif" , "core show function CALLERID" |
| 18:20.10 | bouil | gotoif seems to be what i want. thank you ! |
| 18:20.26 | bouil | i already have the caller id : the callback works |
| 18:20.54 | *** join/#asterisk steliosk (n=Stelios@rouss--1.static.otenet.gr) |
| 18:21.05 | hardwire | hi steliosk |
| 18:25.16 | *** join/#asterisk Mikeonline (i=Mike@unaffiliated/mikeonline) |
| 18:25.27 | Mikeonline | hi |
| 18:26.24 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 18:26.37 | Assid | heya.. who was the one here who works with flowroute? |
| 18:28.41 | Assid | anyone here even tried it? |
| 18:30.06 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 18:32.36 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 18:39.44 | Mikeonline | is there a simple incoming/outgoing/missed call logfile? or only cdr log? |
| 18:40.36 | *** part/#asterisk ChrisN (n=ChrisN@li18-232.members.linode.com) |
| 18:40.55 | *** join/#asterisk wolfelectronic (n=wolfelec@91.112.227.150) |
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| 18:43.03 | *** join/#asterisk afaict (n=afaict@gw2.integral.com.br) |
| 18:43.06 | afaict | Hi folk |
| 18:43.07 | afaict | folks |
| 18:43.19 | LARefugee | lo |
| 18:43.39 | afaict | Are Digium cards able to work in xen virtual machines ? |
| 18:44.37 | afaict | Am I able to use pciback module to hide to xen host the digium cards? |
| 18:44.56 | *** join/#asterisk tobias (n=tobias@user-0c8hj2l.cable.mindspring.com) |
| 18:45.17 | *** join/#asterisk shido6 (n=shido6@209.114.208.111) |
| 18:45.54 | Qwell | afaict: don't use Xen.. |
| 18:46.08 | *** join/#asterisk CrazyTux (n=brandon@nmd.sbx05741.irvinca.wayport.net) |
| 18:46.13 | afaict | Qwell either with xen 3.3.0 ? |
| 18:46.28 | LARefugee | afaict: good question. I doubt dahdi drivers are optimized/virtualized for xen. |
| 18:47.05 | UnixDawg | has never seen a digium card work on a virtual server |
| 18:47.36 | afaict | and ztxen ztdummy modules... doesnt work ? |
| 18:48.15 | UnixDawg | thats zaptel and zaptel is only 1.4 |
| 18:48.20 | UnixDawg | and prev |
| 18:48.29 | UnixDawg | dahdi is the new name |
| 18:48.43 | UnixDawg | and I dont thin xen has been updated for dahdi yet |
| 18:49.00 | UnixDawg | so 1.6 would be a issue |
| 18:49.18 | UnixDawg | zaptel and 1.4 with xtdummy might still work |
| 18:49.19 | *** join/#asterisk steliosk (n=Stelios@91.140.124.241) |
| 18:49.20 | Mikeonline | do i need to change more than just uncommenting the example line in cdr_custom.conf to enable /var/log/asterisk/cdr-custom/Master.csv? |
| 18:49.27 | UnixDawg | but never seen a digium card work |
| 18:49.42 | UnixDawg | on a vps |
| 18:49.50 | UnixDawg | ztdummy is only for timing |
| 18:50.07 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 18:51.21 | afaict | UnixDawg im no body to ask you but whats your experience with xen ? |
| 18:52.14 | UnixDawg | my cousin has been running a server and teaching me |
| 18:52.34 | afaict | UnixDawg :P |
| 18:53.01 | UnixDawg | and we tried to get a tdm400p to work that he has and had no luck |
| 18:53.22 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:53.23 | UnixDawg | he has a single fxs and a single fxs |
| 18:53.44 | afaict | UnixDawg shouldnt digium report xensource.com to write in HCL that digium is not supported for xen ? |
| 18:54.14 | UnixDawg | it supports zaptel ztdummy for timing |
| 18:54.34 | UnixDawg | but zaptel just moved to dahdi for 1.6 so its to new |
| 18:55.13 | UnixDawg | I have not seen a single vps that supports tdm cards |
| 18:55.40 | UnixDawg | not even bsd jails support tdm cards |
| 18:58.13 | tzafrir_laptop | UnixDawg, what's the problem with bsd jails? |
| 18:58.34 | tzafrir_laptop | with linux openvz / vserver the only problem was creating the device file |
| 18:58.37 | jaytee | they don't have cable tv |
| 18:59.52 | UnixDawg | ztdummy works in jails fine but tdm device dont |
| 18:59.57 | UnixDawg | with zaptel |
| 19:00.05 | *** join/#asterisk revengervn (i=60e23f4e@gateway/web/ajax/mibbit.com/x-10344f10cb76e8bf) |
| 19:00.09 | revengervn | hi |
| 19:00.15 | hardwire | whats a vn? |
| 19:00.33 | revengervn | How can I make a conference call with asterisk |
| 19:00.48 | hardwire | your phones support conference calls for 3-way calls |
| 19:00.49 | revengervn | i'm writing a AGI module with python |
| 19:00.52 | hardwire | otherwise you use app_meetme |
| 19:01.05 | jaytee | revengervn, use MeetMe for more than 3 people or use the phone's conference features |
| 19:01.15 | hardwire | jaytee: ahem. |
| 19:01.15 | revengervn | thanks |
| 19:01.21 | hardwire | eyes jaytee |
| 19:01.22 | revengervn | i'm searching it |
| 19:01.25 | jaytee | hardwire, gezundhiet |
| 19:01.30 | hardwire | revengervn: whatchadoin? |
| 19:01.48 | revengervn | actually here is my situation |
| 19:01.50 | hardwire | farmer blows on the back of jaytees head |
| 19:02.00 | revengervn | I'm writing a module for asterisk by using python |
| 19:02.02 | jaytee | farmer? what? |
| 19:02.20 | revengervn | my module will allow a client to call to Asterisk server |
| 19:02.35 | revengervn | asterisk server then call to another fone number |
| 19:02.42 | revengervn | to make a conference call |
| 19:02.47 | revengervn | to put it simply |
| 19:03.07 | revengervn | the client will have conversation with another fone number via our asterisk server |
| 19:03.19 | revengervn | is that possible? |
| 19:03.25 | jaytee | revenger, you're "overthinking" the solution |
| 19:03.40 | revengervn | ... |
| 19:03.54 | revengervn | I can do it another way, can't i ? |
| 19:04.16 | hardwire | 2 parties per conference? |
| 19:04.18 | [TK]D-Fender | revengervn: If you are calling in and * is calling out that isn't even an "conference" |
| 19:04.18 | hardwire | no more? |
| 19:04.29 | [TK]D-Fender | revengervn: thats 1 silly little DIAL command |
| 19:04.44 | revengervn | oh |
| 19:04.49 | hardwire | revengervn: safe face.. quick |
| 19:04.54 | hardwire | tell them your doing it for party line functions |
| 19:05.00 | hardwire | or or .. some other awesome thing |
| 19:05.20 | *** join/#asterisk Defraz (n=T0tal@63.228.246.250) |
| 19:05.43 | petererer | heh |
| 19:05.45 | revengervn | I already tested with Dial cmd |
| 19:05.56 | revengervn | but not successful |
| 19:06.17 | [TK]D-Fender | revengervn: then you did it wrong. |
| 19:06.19 | revengervn | so I was wondering is there another way to achieve that goal |
| 19:06.21 | revengervn | yup |
| 19:06.25 | revengervn | i guess so |
| 19:06.33 | [TK]D-Fender | revengervn: If you can't get DIAL to work, you in more than serious trouble |
| 19:06.40 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-199-127-rrdg-esr-2.dynamic.isadsl.co.za) |
| 19:06.56 | jaytee | revengervn, what determines where Asterisk would try to conference or transfer the call to? |
| 19:06.57 | [TK]D-Fender | revengervn: PASEBIN is your friend, use it. |
| 19:06.58 | [TK]D-Fender | ~pb |
| 19:06.59 | 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 |
| 19:07.01 | *** part/#asterisk Firass-VC22 (n=firass@restek-ws-0.vikcomm.wwu.edu) |
| 19:07.01 | [TK]D-Fender | ^^^^^^^^ |
| 19:07.06 | [TK]D-Fender | jaytee: Don't.... |
| 19:07.13 | [TK]D-Fender | jaytee: Its not a conference. |
| 19:07.19 | jaytee | [TK]D-Fender, I know |
| 19:08.04 | [TK]D-Fender | revengervn: And it might help if you told us how you intend to dial out. What tech? What provider if any? What hardware? VoIP? |
| 19:08.18 | revengervn | jaytee, actually when a client call to our asterisk server, the program will look up a record in database of that client |
| 19:08.32 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:08.45 | revengervn | then determine what fone number our server will call to |
| 19:08.55 | revengervn | i'm using Voip |
| 19:09.09 | jaytee | VOIP? really? wow!!!!! |
| 19:09.18 | jaytee | good choice |
| 19:09.23 | [TK]D-Fender | revengervn: Then go set up your peer properly to dial out. |
| 19:09.38 | revengervn | I registered an account at voicepulse |
| 19:09.42 | jaytee | [TK]D-Fender, from the sound of it it's gonna get uglier than that. |
| 19:09.57 | [TK]D-Fender | jaytee: Oh yeah.... |
| 19:10.31 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 19:10.35 | jaytee | basically sounds like transfer based on database lookup of a client account using callerid because he hasn't mentioned an IVR yet. (shudder) |
| 19:10.45 | revengervn | [TK]D-Fender: I dont understand your idea |
| 19:11.10 | revengervn | yes yaytee |
| 19:11.12 | [TK]D-Fender | revengervn: Idea? this is 1 single DIAL command in the dialplan using your outbound peer. |
| 19:11.45 | [TK]D-Fender | jaytee: forget about selection, I'd be worried that he has no clue how to even place a call out his provider <- |
| 19:12.06 | revengervn | hey [TK]D-FenderI can do it |
| 19:12.11 | jaytee | [TK]D-Fender, yup! hence the ugly part |
| 19:12.17 | [TK]D-Fender | revengervn: You told me you couldn't. Make up your mind. |
| 19:12.49 | revengervn | I could successfully make a call to client |
| 19:13.12 | revengervn | but when a client call to my asterisk, then asterisk call to another fone number |
| 19:13.20 | revengervn | i dont know why i didnt work |
| 19:13.27 | revengervn | it didnt work |
| 19:13.27 | [TK]D-Fender | revengervn: PASTEBIN <- |
| 19:13.47 | [TK]D-Fender | revengervn: you need to look at whats happening at CLI |
| 19:14.07 | Blackvel | guys, sun is shining...ldap thing + ldapget is working :) |
| 19:14.26 | revengervn | :( unfortunately i'm using different computer so I can't paste it out |
| 19:14.43 | revengervn | Fender, I'll try it one more time |
| 19:14.49 | revengervn | thanks a lot for your help |
| 19:14.50 | [TK]D-Fender | revengervn: SSH doesn't care |
| 19:15.36 | revengervn | quick question |
| 19:15.44 | revengervn | can I make a call directly from AGI |
| 19:15.48 | revengervn | but not dial plan? |
| 19:16.29 | [TK]D-Fender | revengervn: AGI = dialplan. |
| 19:16.39 | [TK]D-Fender | revengervn: What is triggering this call? |
| 19:16.57 | revengervn | A client call to our asterisk server |
| 19:17.10 | revengervn | asterisk server will look up his record in database based on his callerid |
| 19:17.36 | revengervn | then asterisk server will determine what number our server should call to |
| 19:17.57 | [TK]D-Fender | revengervn: No need for AGI to dial out, and this is not "direct" |
| 19:18.15 | [TK]D-Fender | revengervn: And yes you can dial in AGI. AGI is just another way to call dialplan apps. |
| 19:18.44 | petererer | Dial is an app. AGI lets you execute apps... |
| 19:18.54 | petererer | 3) profit |
| 19:18.55 | revengervn | but the fone number our server will call is got from database |
| 19:19.14 | [TK]D-Fender | revengervn: that part might be best to have in AGI, but not necessarily |
| 19:19.40 | [TK]D-Fender | revengervn: the actual dial can be after you've done whatever lookup is not viable to do withing the dialplan directly |
| 19:19.43 | *** part/#asterisk spokra (n=spokra@blockhead.sea0.speakeasy.net) |
| 19:19.46 | revengervn | if i remember correctly, there's no way we can get a record from database then place in our dial plan without AGI |
| 19:20.00 | [TK]D-Fender | revengervn: func_odbc , app_mysql, etc |
| 19:20.01 | *** join/#asterisk spokra (n=spokra@blockhead.sea0.speakeasy.net) |
| 19:21.28 | revengervn | thanks Fender |
| 19:21.37 | revengervn | I'm reading it right now |
| 19:22.17 | *** join/#asterisk curtn (n=curtis@cl-451.trn-01.it.sixxs.net) |
| 19:23.25 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 19:24.38 | *** join/#asterisk voxter (n=voxter@mail.metrobridge.com) |
| 19:30.54 | curtn | hi all, does someone use TLS or SRTP in a production environment ? |
| 19:31.15 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 19:32.08 | *** part/#asterisk smth (n=michael@CPE00014e01df2e-CM00195eda2522.cpe.net.cable.rogers.com) |
| 19:35.25 | *** part/#asterisk Mikeonline (i=Mike@unaffiliated/mikeonline) |
| 19:36.02 | *** join/#asterisk nosbig (n=nosbig@cpe-75-185-59-24.columbus.res.rr.com) |
| 19:37.52 | *** join/#asterisk sekil (n=Ognjen@80.93.247.26) |
| 19:45.25 | l2trace99 | anyone know why hangup would spawn another call ? |
| 19:45.37 | voxter | theres something in exten => h, telling it to? |
| 19:45.58 | l2trace99 | no |
| 19:45.58 | gaetronik | i'm looking for ip phone which have hot line feature |
| 19:46.20 | l2trace99 | no exten => h in the context |
| 19:47.13 | gaetronik | like the operator take the phone and the phone call to a num or to s in the dialplan |
| 19:47.22 | *** join/#asterisk saftsack (n=saftsack@ip-77-24-228-184.web.vodafone.de) |
| 19:47.36 | *** join/#asterisk nikolaev (n=nikolaev@93.180.77.179) |
| 19:49.06 | *** join/#asterisk Alan_Hicks (n=alan@cardinal.lizella.net) |
| 19:49.19 | Alan_Hicks | Ok guys. I'm at my wit's end here. |
| 19:50.02 | Alan_Hicks | When I try to dial an outside line, asterisk only dials the first 6 digits of the number and only sends the 7th digit maybe 1 time in every 50. |
| 19:50.10 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 19:50.39 | Alan_Hicks | My dial statement looks like this.... |
| 19:50.41 | Alan_Hicks | exten => _9NXXXXXX,1,Dial(${OUTBOUND_TRUNK}/${EXTEN:1},,T) |
| 19:51.08 | Alan_Hicks | The "9" prefix is not dialed on the Digium card, but the final number entered by the user isn't either. |
| 19:51.25 | Alan_Hicks | If I put an extension like.... |
| 19:51.49 | Alan_Hicks | exten => 287,1,Dial(78495000) ; note- 8 digits, not 7... |
| 19:51.57 | *** join/#asterisk smth (n=michael@CPE00014e01df2e-CM00195eda2522.cpe.net.cable.rogers.com) |
| 19:52.01 | Alan_Hicks | it dials the first seven digits and not the eigth. |
| 19:52.14 | Alan_Hicks | Dial(411) similarly just dials 41. |
| 19:52.24 | UnixDawg | I wish that digium would pull ooh323 out of asterisk-addons |
| 19:52.30 | UnixDawg | I know of no one using it |
| 19:52.35 | UnixDawg | its bloate |
| 19:53.06 | Alan_Hicks | I've confirmed this "missing last digit" with a butt-set and a banjo clip on my line. |
| 19:53.31 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 19:53.38 | Alan_Hicks | Asterisk console claims to dial all the digits, but the butt-set tells a different story. |
| 19:53.39 | *** part/#asterisk bradleyprice86 (n=bprice@68-113-101-90.static.leds.al.charter.com) |
| 19:53.58 | afaict | UnixDawg I have found an australian guy who was able to use asterisk and xen |
| 19:54.03 | *** join/#asterisk Carlos_PHX (n=Carlos@71.36.172.121) |
| 19:54.11 | afaict | UnixDawg with TDP440 |
| 19:54.22 | UnixDawg | did he document it |
| 19:54.29 | UnixDawg | so you can test |
| 19:54.57 | afaict | UnixDawg yes he did document it |
| 19:55.01 | afaict | UnixDawg please see here |
| 19:55.15 | afaict | UnixDawg www.crc.id.au/fedora-core-6-xen-and-asterisk/ |
| 19:55.59 | Knightfal | Guys I am trying to Pass the DID code into the URL parameter exten => s,8,Queue(queue-myqueue|t|DIDHERE||180) But Im not having much luck can anyone assist :) |
| 19:58.03 | afaict | UnixDawg did you see ? |
| 19:58.25 | afaict | UnixDawg i already had experience with XEN and ethernet nic pciback modules. |
| 20:02.17 | Alan_Hicks | No one got any idea? |
| 20:03.23 | *** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 20:05.30 | jaytee | Alan_Hicks, can you pastebin the extensions.conf? |
| 20:05.47 | Alan_Hicks | jaytee: Yeah, it will just take a little while. console access only. |
| 20:05.54 | jaytee | hold on then |
| 20:06.05 | Alan_Hicks | Well, I can just upload the extensions.conf file to my webserver. |
| 20:06.09 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 20:06.13 | hardwire | if I asked nicely do you think LInksys would give me their buildroot for the SPA-94x's |
| 20:06.13 | jaytee | Alan_Hicks, quick question, where are you located? |
| 20:06.15 | hardwire | ? |
| 20:06.17 | hardwire | :) |
| 20:06.20 | Alan_Hicks | jaytee: GA, USA |
| 20:06.35 | jaytee | why are you dialing 8 digits then? |
| 20:06.54 | Alan_Hicks | jaytee: That was just a test to see if it would dial 7 digits if I entered 8. |
| 20:06.57 | hardwire | better than 7 |
| 20:07.31 | UnixDawg | cool |
| 20:07.34 | UnixDawg | yeah |
| 20:07.37 | hardwire | 7++ |
| 20:07.47 | UnixDawg | I forwarded it to my cousin |
| 20:07.57 | UnixDawg | we will see if we can get his card up |
| 20:08.04 | Alan_Hicks | http://www.lizella.net/asterisk/extensions.conf |
| 20:08.12 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:08.22 | Alan_Hicks | Well, that should have worked. Hold on. |
| 20:10.12 | Alan_Hicks | Scratch my last line. I don't know what I was thinking. The link above is my extensions.conf |
| 20:10.29 | tvirus | Is anyone here using Bandwidth.com for SIP trunking and uses the trunks for incoming/outgoing calls? I keep getting dropped audio if both channels are in use and someone attempts to dial out. |
| 20:11.40 | afaict | UnixDawg :) |
| 20:14.02 | *** join/#asterisk thedonvaughn (i=jvaughn@unaffiliated/printk) |
| 20:14.19 | *** join/#asterisk afrosheen (n=afroshee@207.71.49.138) |
| 20:14.24 | UnixDawg | ok bbiab working on dahdi for bsd |
| 20:14.36 | thedonvaughn | Any reason why asterisk can't use 2 different T1 cards with 2 different signallings? one is NI2 and one is DMS100? |
| 20:14.46 | afrosheen | ok, got some weirdness today, I have one voicemail box that doesn't accept a password and is not recording new VMs |
| 20:15.01 | Alan_Hicks | Anyone have any ideas? |
| 20:15.06 | thedonvaughn | I noticed if i plug in a T1 to the DMS100 card, audio cuts in and out. All though the T1 isn't 100% in service, i was just testing the link on the card. |
| 20:15.08 | afaict | Any of you had experience with XEN pciback and TE121? |
| 20:19.07 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 20:21.21 | Knightfal | Anyone have Ideas on passing the DID code in the URL Parameter |
| 20:21.36 | *** join/#asterisk Math` (n=mrene@64.254.252.151) |
| 20:21.45 | Math` | why is asterisk doing DNS lookups on its own generated channel names? |
| 20:21.50 | *** join/#asterisk Meaw (n=dino@b2.86.b6.static.xlhost.com) |
| 20:23.59 | afrosheen | ok, figured out my problem, there was a stale .lock file in my user's INBOX |
| 20:24.03 | *** part/#asterisk nikolaev (n=nikolaev@93.180.77.179) |
| 20:24.39 | Math` | DNS Standard query A 192.168.1.41-b6b01ce8... doesnt make sense |
| 20:25.20 | [TK]D-Fender | Math`: Nice of you to show use the complete output of what caused the error message... |
| 20:26.04 | Math` | its not an error message, asterisk is sending DNS queries targetting its internally generated channel names |
| 20:26.16 | [TK]D-Fender | Math`: Show us where |
| 20:27.31 | Math` | do a "tcpdump udp port 53" on any of your asterisk box |
| 20:27.55 | [TK]D-Fender | Math`: it'd be due to some sort of channel traffic which is debuggable. Go look in CLI |
| 20:28.05 | [TK]D-Fender | but for me here its checkout time. BBIAB |
| 20:31.23 | *** part/#asterisk ftp3 (n=none@pool-96-225-238-78.ptldor.fios.verizon.net) |
| 20:32.11 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 20:32.42 | *** join/#asterisk Firass-VC22 (n=firass@restek-ws-0.vikcomm.wwu.edu) |
| 20:32.55 | rwaite | anyone here have any experiences with aretta communications? |
| 20:35.58 | *** join/#asterisk sdaniels (n=chatzill@216.65.195.52) |
| 20:36.53 | sdaniels | Can someone help me with redirecting an inbound call to an outside number? ex. I want a call to come into an IVR message then route to my cell phone. |
| 20:38.50 | rwaite | sdaniels simply make an extension and place a Dial command in it |
| 20:40.40 | sdaniels | rwaite, this is what I have done, but not working, it does dial an internal extension with no problem, ex. Dial(SIP/6000) works, Dial(SIP/2145555555) does not. outbound calls work fine if called from a physical phone. |
| 20:41.22 | rwaite | sdaniels: pastebin your extensions.conf |
| 20:41.41 | sdaniels | thx, stand by... |
| 20:41.45 | *** part/#asterisk Math` (n=mrene@64.254.252.151) |
| 20:44.23 | sdaniels | [globals] |
| 20:44.24 | sdaniels | VOICEPULSE_API_KEY=####### 27ow6PT2DhEqqadFPGWSrIjMEIYnEGXbi767o7PfPJpmpyT4%3d |
| 20:44.26 | sdaniels | VOICEPULSE_API_PREFIX=VOICEPULSE |
| 20:44.28 | sdaniels | VOICEPULSE_GATEWAY_OUT_A=voicepulse-primary |
| 20:44.29 | sdaniels | VOICEPULSE_GATEWAY_OUT_B=voicepulse-backup |
| 20:44.31 | sdaniels | [general] |
| 20:44.32 | sdaniels | static=yes |
| 20:44.34 | sdaniels | writeprotect=no |
| 20:44.35 | sdaniels | [default] |
| 20:44.37 | sdaniels | [voicepulse-in] |
| 20:44.38 | sdaniels | include => outgoing |
| 20:44.40 | sdaniels | exten => _XX.,1,NoOp(Call received from VoicePulse) |
| 20:44.41 | sdaniels | exten => _XX.,n,Answer() |
| 20:44.43 | sdaniels | exten => _XX.,n,Playback(thank-you-for-calling&tt-allbusy) |
| 20:44.45 | sdaniels | exten => _XX.,n,StartMusicOnHold(default) |
| 20:44.47 | sdaniels | exten => _XX.,n,Wait(10) |
| 20:44.48 | sdaniels | exten => _XX.,n,Dial(SIP/6000,10) |
| 20:44.50 | sdaniels | exten => _XX.,n,Voicemail(6000@default,u) |
| 20:44.51 | sdaniels | exten => _XX.,n,Hangup() |
| 20:44.53 | sdaniels | [banned] |
| 20:44.55 | sdaniels | exten => i,1,Hangup |
| 20:44.56 | sdaniels | exten => t,1,Hangup |
| 20:44.58 | sdaniels | [outgoing] |
| 20:45.00 | sdaniels | exten => _NXXNXXXXXX,1,Set(CALLERID(num)=#######) |
| 20:45.02 | sdaniels | exten => _NXXNXXXXXX,2,Set(CALLERID(name)=NNNN) |
| 20:45.04 | sdaniels | exten => _NXXNXXXXXX,n,NoOp(SIPCALLID: ${SIPCALLID}) |
| 20:45.06 | sdaniels | exten => _NXXNXXXXXX,n,Dial(SIP/+1${EXTEN}@${VOICEPULSE_GATEWAY_OUT_A}) |
| 20:45.08 | sdaniels | exten => _NXXNXXXXXX,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?$1{EXTEN}|GatewayB) |
| 20:45.10 | sdaniels | exten => _NXXNXXXXXX,n(GatewayB),Dial(SIP/+$1{EXTEN}@${VOICEPULSE_GATEWAY_OUT_B} ) |
| 20:45.13 | sdaniels | exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=#######) |
| 20:45.15 | sdaniels | exten => _1NXXNXXXXXX,2,Set(CALLERID(name)=NNNN) |
| 20:45.17 | sdaniels | exten => _1NXXNXXXXXX,n,NoOp(SIPCALLID: ${SIPCALLID}) |
| 20:45.19 | sdaniels | exten => _1NXXNXXXXXX,n,Dial(SIP/+${EXTEN}@${VOICEPULSE_GATEWAY_OUT_A}) |
| 20:45.21 | sdaniels | exten => _1NXXNXXXXXX,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN}|GatewayB) |
| 20:45.23 | sdaniels | exten => _1NXXNXXXXXX,n(GatewayB),Dial(SIP/+${EXTEN}@${VOICEPULSE_GATEWAY_OUT_B} ) |
| 20:45.26 | sdaniels | exten => _011XXXX.,1,Set(CALLERID(num)=0000000000) |
| 20:45.28 | Blackvel | stop |
| 20:45.28 | sdaniels | exten => _011XXXX.,n,NoOp(SIPCALLID: ${SIPCALLID}) |
| 20:45.29 | thehar | oh yeah? i think pastebin is more appropriate for that |
| 20:45.30 | sdaniels | exten => _011XXXX.,n,Dial(SIP/+${EXTEN:3}@${VOICEPULSE_GATEWAY_OUT_A}) |
| 20:45.32 | sdaniels | exten => _011XXXX.,n,GotoIf($[${DIALSTATUS}=CHANUNAVAIL]?${EXTEN}|GatewayB) |
| 20:45.34 | sdaniels | exten => _011XXXX.,n(GatewayB),Dial(SIP/+${EXTEN:3}@${VOICEPULSE_GATEWAY_OUT_B}) |
| 20:45.36 | sdaniels | [phones] |
| 20:45.38 | sdaniels | include => internal |
| 20:45.40 | sdaniels | include => outgoing |
| 20:45.42 | sdaniels | include => ivr |
| 20:45.44 | sdaniels | [ivr] |
| 20:45.46 | sdaniels | exten => 900,1,NoOp(IVR recording test) |
| 20:45.48 | sdaniels | exten => 900,2,Answer() |
| 20:45.50 | sdaniels | exten => 900,3,Background(say-temp-msg-prs-pound) |
| 20:45.52 | sdaniels | exten => 900,4,Record(/var/lib/asterisk/sounds/custom/tempcustommessage.gsm) |
| 20:45.54 | sdaniels | exten => 900,5,Background(to-listen-to-it&press-1,m) |
| 20:45.56 | sdaniels | exten => 900,6,Background(to-accept-recording&press-2,m) |
| 20:45.58 | sdaniels | exten => 900,7,Background(to-rerecord-announce&press-3,m) |
| 20:46.00 | sdaniels | exten => 900,8,WaitExten() |
| 20:46.02 | sdaniels | exten => 1,1,Playback(/var/lib/asterisk/sounds/custom/tempcustommessage) |
| 20:46.04 | sdaniels | exten => 1,2,Goto(900,6) |
| 20:46.06 | sdaniels | exten => 2,1,System(cp /var/lib/asterisk/sounds/custom/tempcustommessage.gsm /va r/lib/asterisk/sounds/custom/ORIGINALMESSAGEname.gsm) |
| 20:46.08 | _ShrikE | oh please |
| 20:46.09 | sdaniels | exten => 2,2,Playback(auth-thankyou) |
| 20:46.11 | sdaniels | exten => 2,3,Hangup() |
| 20:46.13 | sdaniels | exten => 3,1,Goto(900,3) |
| 20:46.15 | sdaniels | [internal] |
| 20:46.17 | sdaniels | exten => 700,1,VoiceMailMain() |
| 20:46.19 | sdaniels | exten => _6XXX,1,NoOp() |
| 20:46.21 | sdaniels | exten => _6XXX,n,Dial(SIP/${EXTEN},10) |
| 20:46.23 | sdaniels | exten => _6XXX,n,Voicemail(${EXTEN}@default,u) |
| 20:46.25 | sdaniels | exten => _6XXX,n,Hangup() |
| 20:46.27 | sdaniels | exten => 62,1,NoOp(Dialling hunt group) |
| 20:46.29 | sdaniels | exten => 62,n,Dial(2142346837,20,t) |
| 20:46.31 | sdaniels | include => outgoing |
| 20:46.33 | sdaniels | [macro-voicepulseflexrate] |
| 20:46.35 | sdaniels | exten => s,1,Set(${VOICEPULSE_API_PREFIX}FLEXRATE=999) |
| 20:46.36 | Blackvel | is he pasting the whole file? wow |
| 20:46.37 | sdaniels | exten => s,2,Set(${VOICEPULSE_API_PREFIX}FLEX_RATE=999) |
| 20:46.39 | sdaniels | exten => s,n,Set(VoicePulsePostData=ApiKey=${ARG1}&PhoneNumber=${ARG2}) |
| 20:46.41 | sdaniels | exten => s,n,Set(VoicePulseResponse=${CURL(https://connect.voicepulse.com/secure /services/Api0605.asmx/AstGetFlexRate|${VoicePulsePostData})}) |
| 20:46.42 | _ShrikE | sdaniels: STOP THAT! |
| 20:46.44 | sdaniels | exten => s,n,Macro(voicepulseparseresponse,${VoicePulseResponse}) |
| 20:46.46 | sdaniels | [macro-voicepulseparseresponse] |
| 20:46.48 | sdaniels | exten => s,1,Set(VoicePulseTemp=${ARG1}) |
| 20:46.50 | sdaniels | exten => s,n,Set(VoicePulseTemp=${CUT(VoicePulseTemp,>,2-)}) |
| 20:46.52 | sdaniels | exten => s,n,Set(VoicePulseTemp=${CUT(VoicePulseTemp,>,2-)}) |
| 20:46.53 | bipolar | he probbly pasted the entire thing already |
| 20:46.54 | sdaniels | exten => s,n,Set(VoicePulseTemp=${CUT(VoicePulseTemp,<,1)}) |
| 20:46.56 | sdaniels | exten => s,n,Set(VoicePulseCounter=${FIELDQTY(VoicePulseTemp,~)}) |
| 20:46.58 | sdaniels | exten => s,n,While($[${VoicePulseCounter} > 0]) |
| 20:47.00 | sdaniels | exten => s,n,Set(VoicePulsePair=${CUT(VoicePulseTemp,~,${VoicePulseCounter})}) |
| 20:47.02 | sdaniels | exten => s,n,Set(VoicePulseKey=${CUT(VoicePulsePair,=,1)}) |
| 20:47.03 | thehar | irssi would have saved us from this |
| 20:47.04 | sdaniels | exten => s,n,Set(VoicePulseValue=${CUT(VoicePulsePair,=,2)}) |
| 20:47.06 | sdaniels | exten => s,n,Set(${VOICEPULSE_API_PREFIX}${VoicePulseKey}=${VoicePulseValue}) |
| 20:47.08 | sdaniels | exten => s,n,Set(VoicePulseCounter=$[${VoicePulseCounter}-1]) |
| 20:47.10 | sdaniels | exten => s,n,EndWhile() |
| 20:47.14 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 20:47.16 | sdaniels | <-- Idiot, sorry about that |
| 20:47.20 | sdaniels | Sorry new to asterisk and IRC, I thought pastebin was a type-o for paste in |
| 20:47.27 | thehar | pastebin.ca |
| 20:47.29 | sdaniels | how do i patebing? |
| 20:47.30 | *** join/#asterisk angryuser (n=Miranda@51.210.20.81.dynamic.adsl.abo.nordnet.fr) |
| 20:47.30 | thehar | use it. love it. |
| 20:47.52 | Meaw | sdaniels, not accepted we know that was deliberated. |
| 20:48.42 | sdaniels | how do I pastebin? |
| 20:48.49 | thehar | http://pastebin.ca |
| 20:48.52 | thehar | ^_^ |
| 20:49.20 | sdaniels | I see, thx. sorry again |
| 20:49.47 | Blackvel | anyways, I am outta here. have a good evening |
| 20:51.01 | sdaniels | Thats alot easier... sorry, anyhow if anyone is still willing to help me after my clear display of stupidity, here is the pastebin link > http://pastebin.ca/1224018 |
| 20:56.25 | angryuser | sdaniels : what is the problem ? |
| 20:56.39 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:57.07 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:58.28 | sdaniels | Other than i just embarrassed my self terribly? I am trying to redirect an incoming call to my cell phone, but I cannot seem to get it to work. the way it is now works dialing ext 6000. if i change 6000 to an outside # the console complains about not being able to find the extension, like its looking for an internal # rather than forwarding it out the trunk. |
| 20:59.33 | jaytee | damn giraffe pulled out one of our outdoor CAT5 cables for one of the webcams. pisses me off they won't bury them or use wireless repeaters |
| 21:01.17 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:02.35 | jaytee | quittin time, be back later |
| 21:04.06 | angryuser | sdaniels : let get it clear, so you re dialing your cell phone in [voicepulse-in] context ? |
| 21:06.54 | sdaniels | yes, the config doesn't reflect that becasue i changed it back to an internal extension for now. |
| 21:08.47 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:10.19 | angryuser | sdaniels : well it is normal that you cell number is not dialed, you have [outgoing] context for dialing out, you should use goto, to redirect your incoming call to you cell phone, there is a lot of ways of doing this, but try chage context and dial out with your outgoing provider, simple |
| 21:10.49 | angryuser | sdaniels : goto, gotoif, whatever |
| 21:11.12 | angryuser | or just dial out directly with your peer |
| 21:12.11 | *** join/#asterisk Carlos_PHX (n=Carlos@71.36.172.121) |
| 21:13.22 | sdaniels | what does the goto command do? just drops the caller into the fist line of the context referenced? |
| 21:13.24 | angryuser | exten => _XX.,n,Dial(SIP/YOURCELLPHONENUMBER@${VOICEPULSE_GATEWAY_OUT_A}) |
| 21:14.04 | gaetronik | !phones |
| 21:14.20 | gaetronik | ~phones |
| 21:14.21 | jbot | it has been said that phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 21:14.40 | angryuser | sdaniels : i prefer to use gotoif and to set a var in astdb |
| 21:15.25 | *** join/#asterisk NirS (n=NirS@80.250.159.240) |
| 21:16.39 | angryuser | sdaniels : and if var=1 >>>redirect enable>>goto [outgoing] to see the syntax of all commands, use CLI core show application goto and core show application gotoif |
| 21:18.04 | sdaniels | I am getting [Oct 9 16:17:05] WARNING[3988]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination), when i user the voicepulsegateway out method |
| 21:18.29 | bouil | did you knwo how to buy one phone number to connect to my sip server , |
| 21:18.30 | bouil | ? |
| 21:19.31 | angryuser | sdaniels : you have exten => _NXXNXXXXXX,n,Dial(SIP/+1${EXTEN}@${VOICEPULSE_GATEWAY_OUT_A}) |
| 21:19.37 | bouil | (DID / DDI number) |
| 21:19.45 | _ShrikE | bouil: you would need to find an itsp |
| 21:19.47 | _ShrikE | ~itsp |
| 21:19.48 | 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 |
| 21:20.10 | angryuser | sdaniels : have you placed +1 prefix ? |
| 21:20.46 | angryuser | sdaniels : can you dial out with this trunk at all ? |
| 21:24.10 | bouil | _ShrikE: thanks |
| 21:24.13 | bouil | bye |
| 21:29.40 | *** join/#asterisk hfb (n=hfb@pool-98-112-220-30.lsanca.dsl-w.verizon.net) |
| 21:30.30 | *** join/#asterisk Greek-Boy (n=email@41.221.58.13) |
| 21:30.47 | *** join/#asterisk saftsack (n=saftsack@ip-77-25-41-74.web.vodafone.de) |
| 21:30.49 | angryuser | btw i have found a job related to asterisk hourra ;) |
| 21:31.42 | angryuser | or i should better say, asterisk helped me to find a job |
| 21:31.48 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 21:33.29 | [netman] | congratulations angryuser |
| 21:42.07 | *** join/#asterisk gcbirzan (n=gcbirzan@pida/gcbirzan) |
| 21:43.58 | gaetronik | ~phones |
| 21:43.58 | jbot | i guess phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 21:44.16 | gaetronik | to get the end of the message? |
| 21:58.04 | *** part/#asterisk Alan_Hicks (n=alan@cardinal.lizella.net) |
| 22:04.19 | *** join/#asterisk hi365_m (n=hi365@213.151.52.225) |
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| 22:06.40 | *** join/#asterisk javb (n=voip1@tdev212-131.codetel.net.do) |
| 22:07.41 | javb | Guys, i've done an IAX trunk. Calls from one PBX to another gives "No authority found" problem. But, adding, in iax.conf, an account to the pbx itself, problem solved... Any idea why this? |
| 22:10.37 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
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| 22:39.29 | *** join/#asterisk kernel_ghost (n=propagan@CPE-121-223-206-243.static.nsw.bigpond.net.au) |
| 22:40.15 | kernel_ghost | hey people, I have a new install running asterisk 1.4 branch, and all the system recordings run really slow and laggy, sounds very entertaining but not nice!! anything I can look at? |
| 22:40.23 | kernel_ghost | the system has plenty of grunt too |
| 22:42.28 | voxter | really? theres pretty much been no actual talk in here today? just joins/parts?? |
| 22:42.40 | voxter | or at least for the past 3 hours |
| 22:46.09 | *** part/#asterisk foamball (n=eisert@user-0c6sp2h.cable.mindspring.com) |
| 22:48.00 | *** part/#asterisk patrick-- (n=patrick@212.112.255.47) |
| 22:50.27 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:51.11 | *** join/#asterisk CGMChris (n=chris@mail.cgmyes.com) |
| 22:51.32 | *** join/#asterisk Carlos_PHX (n=Carlos@71.36.172.121) |
| 22:51.42 | CGMChris | Has anybody pieced together the default sound files to make a message to the effect of "we are closed for the holiday", or "we are closed july 4", "currently unavail for july 4", etc. ? |
| 22:52.10 | CGMChris | If it's possible, I'm sure not putting the pieces of the puzzle together in the correct order. |
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| 23:31.04 | TekkenWar | any1 willing help me about setting up a pbx server? |
| 23:33.33 | CGMChris | Anyones eyeballs ever bled? I think mine are close. |
| 23:33.47 | CGMChris | TekkenWar: I won't do it for you, but; What's your question? |
| 23:34.39 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 23:35.10 | x86 | is there a card in between an 8 port and 24 port? |
| 23:36.00 | coppice | since you can progressively populate them, would such a card be a good thing? |
| 23:36.07 | x86 | hmm, guess I can get a 24-port card and configure 12 analog stations (3x4FXS modules) and 8 CO lines (2x4FXO modules) |
| 23:36.17 | x86 | yeah |
| 23:36.21 | coppice | some of he Chinese make 12 channel cards |
| 23:36.38 | x86 | I've got a need for 10 phones and 7 CO lines |
| 23:36.50 | x86 | wanted to do it with the least amount of overkill ;) |
| 23:40.16 | *** part/#asterisk jpastore (n=jpastore@69.65.65.40) |
| 23:40.18 | x86 | 1TDM2432EF will do the trick, eh? |
| 23:45.31 | ManxPower | kernel_ghost: stop asterisk then rmmod ztdummy then start asterisk. If the problem goes away your kernel HZ is not 1000 |
| 23:45.58 | kernel_ghost | ManxPower: thanks heaps, sorry to be ignorant but what does that mean the kernel hz not 1000 |
| 23:46.19 | kernel_ghost | and if that fixes it, do i just stop ztdummy from being loaded? |
| 23:46.22 | ManxPower | kernel_ghost: It's some compile setting in the Linux kernel having to do with the RTC |
| 23:46.54 | ManxPower | kernel_ghost: you need ztdummy for meetme and iax2 call header compression (called IAX2 trunking). If you don't need either of those just don't load zaptel |
| 23:47.12 | kernel_ghost | ManxPower: sweet I dont need them! |
| 23:47.15 | CGMChris | ztdummy keeps track of time in Asterisk. kernel hz refers to the speed at which events happen... Low level OS systems like application managers and such. |
| 23:47.52 | ManxPower | ztdummy does not keep track of time in asterisk |
| 23:48.42 | CGMChris | Incorrect, quote the book, page 28: ztdummy is a clocking mechanism designed to provide a timing source to a system where no hardware timing source exists. |
| 23:49.09 | ManxPower | ztdummy provides the sync (some people call this a timer but it is more like a really really rally fast metronome, it has no concept of seconds only ticks) and audio mixing. |
| 23:49.25 | ManxPower | CGMChris: and that has nothing to do with "time" |
| 23:49.37 | CGMChris | ticks relate directly to time based on frequency |
| 23:50.01 | ManxPower | CGMChris: my point is that ztdummy has nothing to do with calendar time. |
| 23:50.08 | CGMChris | Sure, I'll give you that. |
| 23:50.53 | kernel_ghost | ManxPower: ok |
| 23:50.59 | kernel_ghost | that seemed to fix it |
| 23:51.00 | ManxPower | now if you want to call something that does nothing but tick 1000 times per second a "clock" or "something that keeps track of time"..... |
| 23:51.19 | CGMChris | You'll have to forgive me, I tend to dumb down my explanations as if I am talking to a plumber (probably because most of my clients are). I will try to use more precise technical terms in this room. |
| 23:51.23 | ManxPower | in telco terms "clock" means "sync source" |
| 23:51.53 | coppice | i thought clock was the lunch and home time indicator |
| 23:51.56 | ManxPower | CGMChris: call it a clock and people try to install ntpd to fix the problem |
| 23:52.39 | CGMChris | ManxPower: That may not be a bad thing... at least their time would be in sync with the rest of the world. |
| 23:53.43 | *** join/#asterisk LiNeTuX_Home (n=LiNeTuX@253.238.95.24.cfl.res.rr.com) |
| 23:54.07 | ManxPower | I don't even bother to correct misinformation on the asterisk-users mailing list anymore. It would be a full time job. |
| 23:54.15 | *** join/#asterisk seanmh (n=seanmh@c-68-35-21-64.hsd1.nm.comcast.net) |
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| 23:55.33 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 23:57.17 | kernel_ghost | manxpower, how can I tell zaptel not to load |
| 23:57.19 | kernel_ghost | at boot now |
| 23:58.18 | CGMChris | I have a magnificant 154 line extensions.conf and 85 line sip.conf.... MUCH smaller than the gui provided. |
| 23:58.36 | CGMChris | took MUCH longer than 154 lines of code in any programming language i've ever used. |
